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<span class="pre noprint docinfo">[<a href="https://www.rfc-editor.org" title="RFC Editor">RFC Home</a>] [<a href="/rfc/rfc2543.txt">TEXT</a>|<a href="/rfc/pdfrfc/rfc2543.txt.pdf">PDF</a>|<a href="/rfc/rfc2543.html">HTML</a>] [<a href='https://datatracker.ietf.org/doc/rfc2543' title='IETF Datatracker information for this document'>Tracker</a>] [<a href="https://datatracker.ietf.org/ipr/search/?rfc=2543&amp;submit=rfc" title="IPR disclosures related to this document">IPR</a>] [<a href='https://www.rfc-editor.org/info/rfc2543' title='Info page'>Info page</a>]                  </span><br/><span class="pre noprint docinfo">                                                                        </span><br /><span class="pre noprint docinfo">Obsoleted by: <a href="/rfc/rfc3261" target="_blank">3261</a>, <a href="/rfc/rfc3262" target="_blank">3262</a>, <a href="/rfc/rfc3263" target="_blank">3263</a>, <a href="/rfc/rfc3264" target="_blank">3264</a>, <a href="/rfc/rfc3265" target="_blank">3265</a>             PROPOSED STANDARD</span><br /><span class="pre noprint docinfo">                                                                        </span><pre>Network Working Group                                          M. Handley
Request for Comments: 2543                                          ACIRI
Category: Standards Track                                  H. Schulzrinne
                                                              Columbia U.
                                                              E. Schooler
                                                                 Cal Tech
                                                             J. Rosenberg
                                                                Bell Labs
                                                               March 1999

                    <span class="h1">SIP: Session Initiation Protocol</span>

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (1999).  All Rights Reserved.

IESG Note

   The IESG intends to charter, in the near future, one or more working
   groups to produce standards for "name lookup", where such names would
   include electronic mail addresses and telephone numbers, and the
   result of such a lookup would be a list of attributes and
   characteristics of the user or terminal associated with the name.
   Groups which are in need of a "name lookup" protocol should follow
   the development of these new working groups rather than using SIP for
   this function. In addition it is anticipated that SIP will migrate
   towards using such protocols, and SIP implementors are advised to
   monitor these efforts.

Abstract

   The Session Initiation Protocol (SIP) is an application-layer control
   (signaling) protocol for creating, modifying and terminating sessions
   with one or more participants. These sessions include Internet
   multimedia conferences, Internet telephone calls and multimedia
   distribution. Members in a session can communicate via multicast or
   via a mesh of unicast relations, or a combination of these.






<span class="grey">Handley, et al.             Standards Track                     [Page 1]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-2" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   SIP invitations used to create sessions carry session descriptions
   which allow participants to agree on a set of compatible media types.
   SIP supports user mobility by proxying and redirecting requests to
   the user's current location. Users can register their current
   location.  SIP is not tied to any particular conference control
   protocol. SIP is designed to be independent of the lower-layer
   transport protocol and can be extended with additional capabilities.

Table of Contents

   <a href="#section-1">1</a>          Introduction ........................................    <a href="#page-7">7</a>
   <a href="#section-1.1">1.1</a>        Overview of SIP Functionality .......................    <a href="#page-7">7</a>
   <a href="#section-1.2">1.2</a>        Terminology .........................................    <a href="#page-8">8</a>
   <a href="#section-1.3">1.3</a>        Definitions .........................................    <a href="#page-9">9</a>
   <a href="#section-1.4">1.4</a>        Overview of SIP Operation ...........................   <a href="#page-12">12</a>
   <a href="#section-1.4.1">1.4.1</a>      SIP Addressing ......................................   <a href="#page-12">12</a>
   <a href="#section-1.4.2">1.4.2</a>      Locating a SIP Server ...............................   <a href="#page-13">13</a>
   <a href="#section-1.4.3">1.4.3</a>      SIP Transaction .....................................   <a href="#page-14">14</a>
   <a href="#section-1.4.4">1.4.4</a>      SIP Invitation ......................................   <a href="#page-15">15</a>
   <a href="#section-1.4.5">1.4.5</a>      Locating a User .....................................   <a href="#page-17">17</a>
   <a href="#section-1.4.6">1.4.6</a>      Changing an Existing Session ........................   <a href="#page-18">18</a>
   <a href="#section-1.4.7">1.4.7</a>      Registration Services ...............................   <a href="#page-18">18</a>
   <a href="#section-1.5">1.5</a>        Protocol Properties .................................   <a href="#page-18">18</a>
   <a href="#section-1.5.1">1.5.1</a>      Minimal State .......................................   <a href="#page-18">18</a>
   <a href="#section-1.5.2">1.5.2</a>      Lower-Layer-Protocol Neutral ........................   <a href="#page-18">18</a>
   <a href="#section-1.5.3">1.5.3</a>      Text-Based ..........................................   <a href="#page-20">20</a>
   <a href="#section-2">2</a>          SIP Uniform Resource Locators .......................   <a href="#page-20">20</a>
   <a href="#section-3">3</a>          SIP Message Overview ................................   <a href="#page-24">24</a>
   <a href="#section-4">4</a>          Request .............................................   <a href="#page-26">26</a>
   <a href="#section-4.1">4.1</a>        Request-Line ........................................   <a href="#page-26">26</a>
   <a href="#section-4.2">4.2</a>        Methods .............................................   <a href="#page-27">27</a>
   <a href="#section-4.2.1">4.2.1</a>      INVITE ..............................................   <a href="#page-28">28</a>
   <a href="#section-4.2.2">4.2.2</a>      ACK .................................................   <a href="#page-29">29</a>
   <a href="#section-4.2.3">4.2.3</a>      OPTIONS .............................................   <a href="#page-29">29</a>
   <a href="#section-4.2.4">4.2.4</a>      BYE .................................................   <a href="#page-30">30</a>
   <a href="#section-4.2.5">4.2.5</a>      CANCEL ..............................................   <a href="#page-30">30</a>
   <a href="#section-4.2.6">4.2.6</a>      REGISTER ............................................   <a href="#page-31">31</a>
   <a href="#section-4.3">4.3</a>        Request-URI .........................................   <a href="#page-34">34</a>
   <a href="#section-4.3.1">4.3.1</a>      SIP Version .........................................   <a href="#page-35">35</a>
   <a href="#section-4.4">4.4</a>        Option Tags .........................................   <a href="#page-35">35</a>
   <a href="#section-4.4.1">4.4.1</a>      Registering New Option Tags with IANA ...............   <a href="#page-35">35</a>
   <a href="#section-5">5</a>          Response ............................................   <a href="#page-36">36</a>
   <a href="#section-5.1">5.1</a>        Status-Line .........................................   <a href="#page-36">36</a>
   <a href="#section-5.1.1">5.1.1</a>      Status Codes and Reason Phrases .....................   <a href="#page-37">37</a>
   <a href="#section-6">6</a>          Header Field Definitions ............................   <a href="#page-39">39</a>
   <a href="#section-6.1">6.1</a>        General Header Fields ...............................   <a href="#page-41">41</a>
   <a href="#section-6.2">6.2</a>        Entity Header Fields ................................   <a href="#page-42">42</a>
   <a href="#section-6.3">6.3</a>        Request Header Fields ...............................   <a href="#page-43">43</a>



<span class="grey">Handley, et al.             Standards Track                     [Page 2]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-3" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   <a href="#section-6.4">6.4</a>        Response Header Fields ..............................   <a href="#page-43">43</a>
   <a href="#section-6.5">6.5</a>        End-to-end and Hop-by-hop Headers ...................   <a href="#page-43">43</a>
   <a href="#section-6.6">6.6</a>        Header Field Format .................................   <a href="#page-43">43</a>
   <a href="#section-6.7">6.7</a>        Accept ..............................................   <a href="#page-44">44</a>
   <a href="#section-6.8">6.8</a>        Accept-Encoding .....................................   <a href="#page-44">44</a>
   <a href="#section-6.9">6.9</a>        Accept-Language .....................................   <a href="#page-45">45</a>
   <a href="#section-6.10">6.10</a>       Allow ...............................................   <a href="#page-45">45</a>
   <a href="#section-6.11">6.11</a>       Authorization .......................................   <a href="#page-45">45</a>
   <a href="#section-6.12">6.12</a>       Call-ID .............................................   <a href="#page-46">46</a>
   <a href="#section-6.13">6.13</a>       Contact .............................................   <a href="#page-47">47</a>
   <a href="#section-6.14">6.14</a>       Content-Encoding ....................................   <a href="#page-50">50</a>
   <a href="#section-6.15">6.15</a>       Content-Length ......................................   <a href="#page-51">51</a>
   <a href="#section-6.16">6.16</a>       Content-Type ........................................   <a href="#page-51">51</a>
   <a href="#section-6.17">6.17</a>       CSeq ................................................   <a href="#page-52">52</a>
   <a href="#section-6.18">6.18</a>       Date ................................................   <a href="#page-53">53</a>
   <a href="#section-6.19">6.19</a>       Encryption ..........................................   <a href="#page-54">54</a>
   <a href="#section-6.20">6.20</a>       Expires .............................................   <a href="#page-55">55</a>
   <a href="#section-6.21">6.21</a>       From ................................................   <a href="#page-56">56</a>
   <a href="#section-6.22">6.22</a>       Hide ................................................   <a href="#page-57">57</a>
   <a href="#section-6.23">6.23</a>       Max-Forwards ........................................   <a href="#page-59">59</a>
   <a href="#section-6.24">6.24</a>       Organization ........................................   <a href="#page-59">59</a>
   <a href="#section-6.25">6.25</a>       Priority ............................................   <a href="#page-60">60</a>
   <a href="#section-6.26">6.26</a>       Proxy-Authenticate ..................................   <a href="#page-60">60</a>
   <a href="#section-6.27">6.27</a>       Proxy-Authorization .................................   <a href="#page-61">61</a>
   <a href="#section-6.28">6.28</a>       Proxy-Require .......................................   <a href="#page-61">61</a>
   <a href="#section-6.29">6.29</a>       Record-Route ........................................   <a href="#page-62">62</a>
   <a href="#section-6.30">6.30</a>       Require .............................................   <a href="#page-63">63</a>
   <a href="#section-6.31">6.31</a>       Response-Key ........................................   <a href="#page-63">63</a>
   <a href="#section-6.32">6.32</a>       Retry-After .........................................   <a href="#page-64">64</a>
   <a href="#section-6.33">6.33</a>       Route ...............................................   <a href="#page-65">65</a>
   <a href="#section-6.34">6.34</a>       Server ..............................................   <a href="#page-65">65</a>
   <a href="#section-6.35">6.35</a>       Subject .............................................   <a href="#page-65">65</a>
   <a href="#section-6.36">6.36</a>       Timestamp ...........................................   <a href="#page-66">66</a>
   <a href="#section-6.37">6.37</a>       To ..................................................   <a href="#page-66">66</a>
   <a href="#section-6.38">6.38</a>       Unsupported .........................................   <a href="#page-68">68</a>
   <a href="#section-6.39">6.39</a>       User-Agent ..........................................   <a href="#page-68">68</a>
   <a href="#section-6.40">6.40</a>       Via .................................................   <a href="#page-68">68</a>
   <a href="#section-6.40.1">6.40.1</a>     Requests ............................................   <a href="#page-68">68</a>
   <a href="#section-6.40.2">6.40.2</a>     Receiver-tagged Via Header Fields ...................   <a href="#page-69">69</a>
   <a href="#section-6.40.3">6.40.3</a>     Responses ...........................................   <a href="#page-70">70</a>
   <a href="#section-6.40.4">6.40.4</a>     User Agent and Redirect Servers .....................   <a href="#page-70">70</a>
   <a href="#section-6.40.5">6.40.5</a>     Syntax ..............................................   <a href="#page-71">71</a>
   <a href="#section-6.41">6.41</a>       Warning .............................................   <a href="#page-72">72</a>
   <a href="#section-6.42">6.42</a>       WWW-Authenticate ....................................   <a href="#page-74">74</a>
   <a href="#section-7">7</a>          Status Code Definitions .............................   <a href="#page-75">75</a>
   <a href="#section-7.1">7.1</a>        Informational 1xx ...................................   <a href="#page-75">75</a>
   <a href="#section-7.1.1">7.1.1</a>      100 Trying ..........................................   <a href="#page-75">75</a>
   <a href="#section-7.1.2">7.1.2</a>      180 Ringing .........................................   <a href="#page-75">75</a>



<span class="grey">Handley, et al.             Standards Track                     [Page 3]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-4" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   <a href="#section-7.1.3">7.1.3</a>      181 Call Is Being Forwarded .........................   <a href="#page-75">75</a>
   <a href="#section-7.1.4">7.1.4</a>      182 Queued ..........................................   <a href="#page-76">76</a>
   <a href="#section-7.2">7.2</a>        Successful 2xx ......................................   <a href="#page-76">76</a>
   <a href="#section-7.2.1">7.2.1</a>      200 OK ..............................................   <a href="#page-76">76</a>
   <a href="#section-7.3">7.3</a>        Redirection 3xx .....................................   <a href="#page-76">76</a>
   <a href="#section-7.3.1">7.3.1</a>      300 Multiple Choices ................................   <a href="#page-77">77</a>
   <a href="#section-7.3.2">7.3.2</a>      301 Moved Permanently ...............................   <a href="#page-77">77</a>
   <a href="#section-7.3.3">7.3.3</a>      302 Moved Temporarily ...............................   <a href="#page-77">77</a>
   <a href="#section-7.3.4">7.3.4</a>      305 Use Proxy .......................................   <a href="#page-77">77</a>
   <a href="#section-7.3.5">7.3.5</a>      380 Alternative Service .............................   <a href="#page-78">78</a>
   <a href="#section-7.4">7.4</a>        Request Failure 4xx .................................   <a href="#page-78">78</a>
   <a href="#section-7.4.1">7.4.1</a>      400 Bad Request .....................................   <a href="#page-78">78</a>
   <a href="#section-7.4.2">7.4.2</a>      401 Unauthorized ....................................   <a href="#page-78">78</a>
   <a href="#section-7.4.3">7.4.3</a>      402 Payment Required ................................   <a href="#page-78">78</a>
   <a href="#section-7.4.4">7.4.4</a>      403 Forbidden .......................................   <a href="#page-78">78</a>
   <a href="#section-7.4.5">7.4.5</a>      404 Not Found .......................................   <a href="#page-78">78</a>
   <a href="#section-7.4.6">7.4.6</a>      405 Method Not Allowed ..............................   <a href="#page-78">78</a>
   <a href="#section-7.4.7">7.4.7</a>      406 Not Acceptable ..................................   <a href="#page-79">79</a>
   <a href="#section-7.4.8">7.4.8</a>      407 Proxy Authentication Required ...................   <a href="#page-79">79</a>
   <a href="#section-7.4.9">7.4.9</a>      408 Request Timeout .................................   <a href="#page-79">79</a>
   <a href="#section-7.4.10">7.4.10</a>     409 Conflict ........................................   <a href="#page-79">79</a>
   <a href="#section-7.4.11">7.4.11</a>     410 Gone ............................................   <a href="#page-79">79</a>
   <a href="#section-7.4.12">7.4.12</a>     411 Length Required .................................   <a href="#page-79">79</a>
   <a href="#section-7.4.13">7.4.13</a>     413 Request Entity Too Large ........................   <a href="#page-80">80</a>
   <a href="#section-7.4.14">7.4.14</a>     414 Request-URI Too Long ............................   <a href="#page-80">80</a>
   <a href="#section-7.4.15">7.4.15</a>     415 Unsupported Media Type ..........................   <a href="#page-80">80</a>
   <a href="#section-7.4.16">7.4.16</a>     420 Bad Extension ...................................   <a href="#page-80">80</a>
   <a href="#section-7.4.17">7.4.17</a>     480 Temporarily Unavailable .........................   <a href="#page-80">80</a>
   <a href="#section-7.4.18">7.4.18</a>     481 Call Leg/Transaction Does Not Exist .............   <a href="#page-81">81</a>
   <a href="#section-7.4.19">7.4.19</a>     482 Loop Detected ...................................   <a href="#page-81">81</a>
   <a href="#section-7.4.20">7.4.20</a>     483 Too Many Hops ...................................   <a href="#page-81">81</a>
   <a href="#section-7.4.21">7.4.21</a>     484 Address Incomplete ..............................   <a href="#page-81">81</a>
   <a href="#section-7.4.22">7.4.22</a>     485 Ambiguous .......................................   <a href="#page-81">81</a>
   <a href="#section-7.4.23">7.4.23</a>     486 Busy Here .......................................   <a href="#page-82">82</a>
   <a href="#section-7.5">7.5</a>        Server Failure 5xx ..................................   <a href="#page-82">82</a>
   <a href="#section-7.5.1">7.5.1</a>      500 Server Internal Error ...........................   <a href="#page-82">82</a>
   <a href="#section-7.5.2">7.5.2</a>      501 Not Implemented .................................   <a href="#page-82">82</a>
   <a href="#section-7.5.3">7.5.3</a>      502 Bad Gateway .....................................   <a href="#page-82">82</a>
   <a href="#section-7.5.4">7.5.4</a>      503 Service Unavailable .............................   <a href="#page-83">83</a>
   <a href="#section-7.5.5">7.5.5</a>      504 Gateway Time-out ................................   <a href="#page-83">83</a>
   <a href="#section-7.5.6">7.5.6</a>      505 Version Not Supported ...........................   <a href="#page-83">83</a>
   <a href="#section-7.6">7.6</a>        Global Failures 6xx .................................   <a href="#page-83">83</a>
   <a href="#section-7.6.1">7.6.1</a>      600 Busy Everywhere .................................   <a href="#page-83">83</a>
   <a href="#section-7.6.2">7.6.2</a>      603 Decline .........................................   <a href="#page-84">84</a>
   <a href="#section-7.6.3">7.6.3</a>      604 Does Not Exist Anywhere .........................   <a href="#page-84">84</a>
   <a href="#section-7.6.4">7.6.4</a>      606 Not Acceptable ..................................   <a href="#page-84">84</a>
   <a href="#section-8">8</a>          SIP Message Body ....................................   <a href="#page-84">84</a>
   <a href="#section-8.1">8.1</a>        Body Inclusion ......................................   <a href="#page-84">84</a>



<span class="grey">Handley, et al.             Standards Track                     [Page 4]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-5" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   <a href="#section-8.2">8.2</a>        Message Body Type ...................................   <a href="#page-85">85</a>
   <a href="#section-8.3">8.3</a>        Message Body Length .................................   <a href="#page-85">85</a>
   <a href="#section-9">9</a>          Compact Form ........................................   <a href="#page-85">85</a>
   <a href="#section-10">10</a>         Behavior of SIP Clients and Servers .................   <a href="#page-86">86</a>
   <a href="#section-10.1">10.1</a>       General Remarks .....................................   <a href="#page-86">86</a>
   <a href="#section-10.1.1">10.1.1</a>     Requests ............................................   <a href="#page-86">86</a>
   <a href="#section-10.1.2">10.1.2</a>     Responses ...........................................   <a href="#page-87">87</a>
   10.2       Source Addresses, Destination Addresses and
              Connections .........................................   <a href="#page-88">88</a>
   <a href="#section-10.2.1">10.2.1</a>     Unicast UDP .........................................   <a href="#page-88">88</a>
   <a href="#section-10.2.2">10.2.2</a>     Multicast UDP .......................................   <a href="#page-88">88</a>
   <a href="#section-10.3">10.3</a>       TCP .................................................   <a href="#page-89">89</a>
   10.4       Reliability for BYE, CANCEL, OPTIONS, REGISTER
              Requests ............................................   <a href="#page-90">90</a>
   <a href="#section-10.4.1">10.4.1</a>     UDP .................................................   <a href="#page-90">90</a>
   <a href="#section-10.4.2">10.4.2</a>     TCP .................................................   <a href="#page-91">91</a>
   <a href="#section-10.5">10.5</a>       Reliability for INVITE Requests .....................   <a href="#page-91">91</a>
   <a href="#section-10.5.1">10.5.1</a>     UDP .................................................   <a href="#page-92">92</a>
   <a href="#section-10.5.2">10.5.2</a>     TCP .................................................   <a href="#page-95">95</a>
   <a href="#section-10.6">10.6</a>       Reliability for ACK Requests ........................   <a href="#page-95">95</a>
   <a href="#section-10.7">10.7</a>       ICMP Handling .......................................   <a href="#page-95">95</a>
   <a href="#section-11">11</a>         Behavior of SIP User Agents .........................   <a href="#page-95">95</a>
   <a href="#section-11.1">11.1</a>       Caller Issues Initial INVITE Request ................   <a href="#page-96">96</a>
   <a href="#section-11.2">11.2</a>       Callee Issues Response ..............................   <a href="#page-96">96</a>
   <a href="#section-11.3">11.3</a>       Caller Receives Response to Initial Request .........   <a href="#page-96">96</a>
   <a href="#section-11.4">11.4</a>       Caller or Callee Generate Subsequent Requests .......   <a href="#page-97">97</a>
   <a href="#section-11.5">11.5</a>       Receiving Subsequent Requests .......................   <a href="#page-97">97</a>
   <a href="#section-12">12</a>         Behavior of SIP Proxy and Redirect Servers ..........   <a href="#page-97">97</a>
   <a href="#section-12.1">12.1</a>       Redirect Server .....................................   <a href="#page-97">97</a>
   <a href="#section-12.2">12.2</a>       User Agent Server ...................................   <a href="#page-98">98</a>
   <a href="#section-12.3">12.3</a>       Proxy Server ........................................   <a href="#page-98">98</a>
   <a href="#section-12.3.1">12.3.1</a>     Proxying Requests ...................................   <a href="#page-98">98</a>
   <a href="#section-12.3.2">12.3.2</a>     Proxying Responses ..................................   <a href="#page-99">99</a>
   <a href="#section-12.3.3">12.3.3</a>     Stateless Proxy: Proxying Responses .................   <a href="#page-99">99</a>
   <a href="#section-12.3.4">12.3.4</a>     Stateful Proxy: Receiving Requests ..................   <a href="#page-99">99</a>
   <a href="#section-12.3.5">12.3.5</a>     Stateful Proxy: Receiving ACKs ......................   <a href="#page-99">99</a>
   <a href="#section-12.3.6">12.3.6</a>     Stateful Proxy: Receiving Responses .................  <a href="#page-100">100</a>
   <a href="#section-12.3.7">12.3.7</a>     Stateless, Non-Forking Proxy ........................  <a href="#page-100">100</a>
   <a href="#section-12.4">12.4</a>       Forking Proxy .......................................  <a href="#page-100">100</a>
   <a href="#section-13">13</a>         Security Considerations .............................  <a href="#page-104">104</a>
   <a href="#section-13.1">13.1</a>       Confidentiality and Privacy: Encryption .............  <a href="#page-104">104</a>
   <a href="#section-13.1.1">13.1.1</a>     End-to-End Encryption ...............................  <a href="#page-104">104</a>
   <a href="#section-13.1.2">13.1.2</a>     Privacy of SIP Responses ............................  <a href="#page-107">107</a>
   <a href="#section-13.1.3">13.1.3</a>     Encryption by Proxies ...............................  <a href="#page-108">108</a>
   <a href="#section-13.1.4">13.1.4</a>     Hop-by-Hop Encryption ...............................  <a href="#page-108">108</a>
   <a href="#section-13.1.5">13.1.5</a>     Via field encryption ................................  <a href="#page-108">108</a>
   13.2       Message Integrity and Access Control:
              Authentication ......................................  <a href="#page-109">109</a>



<span class="grey">Handley, et al.             Standards Track                     [Page 5]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-6" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   <a href="#section-13.2.1">13.2.1</a>     Trusting responses ..................................  <a href="#page-112">112</a>
   <a href="#section-13.3">13.3</a>       Callee Privacy ......................................  <a href="#page-113">113</a>
   <a href="#section-13.4">13.4</a>       Known Security Problems .............................  <a href="#page-113">113</a>
   14         SIP Authentication using HTTP Basic and Digest
              Schemes .............................................  <a href="#page-113">113</a>
   <a href="#section-14.1">14.1</a>       Framework ...........................................  <a href="#page-113">113</a>
   <a href="#section-14.2">14.2</a>       Basic Authentication ................................  <a href="#page-114">114</a>
   <a href="#section-14.3">14.3</a>       Digest Authentication ...............................  <a href="#page-114">114</a>
   <a href="#section-14.4">14.4</a>       Proxy-Authentication ................................  <a href="#page-115">115</a>
   <a href="#section-15">15</a>         SIP Security Using PGP ..............................  <a href="#page-115">115</a>
   <a href="#section-15.1">15.1</a>       PGP Authentication Scheme ...........................  <a href="#page-115">115</a>
   <a href="#section-15.1.1">15.1.1</a>     The WWW-Authenticate Response Header ................  <a href="#page-116">116</a>
   <a href="#section-15.1.2">15.1.2</a>     The Authorization Request Header ....................  <a href="#page-117">117</a>
   <a href="#section-15.2">15.2</a>       PGP Encryption Scheme ...............................  <a href="#page-118">118</a>
   <a href="#section-15.3">15.3</a>       Response-Key Header Field for PGP ...................  <a href="#page-119">119</a>
   <a href="#section-16">16</a>         Examples ............................................  <a href="#page-119">119</a>
   <a href="#section-16.1">16.1</a>       Registration ........................................  <a href="#page-119">119</a>
   <a href="#section-16.2">16.2</a>       Invitation to a Multicast Conference ................  <a href="#page-121">121</a>
   <a href="#section-16.2.1">16.2.1</a>     Request .............................................  <a href="#page-121">121</a>
   <a href="#section-16.2.2">16.2.2</a>     Response ............................................  <a href="#page-122">122</a>
   <a href="#section-16.3">16.3</a>       Two-party Call ......................................  <a href="#page-123">123</a>
   <a href="#section-16.4">16.4</a>       Terminating a Call ..................................  <a href="#page-125">125</a>
   <a href="#section-16.5">16.5</a>       Forking Proxy .......................................  <a href="#page-126">126</a>
   <a href="#section-16.6">16.6</a>       Redirects ...........................................  <a href="#page-130">130</a>
   <a href="#section-16.7">16.7</a>       Negotiation .........................................  <a href="#page-131">131</a>
   <a href="#section-16.8">16.8</a>       OPTIONS Request .....................................  <a href="#page-132">132</a>
   <a href="#appendix-A">A</a>          Minimal Implementation ..............................  <a href="#page-134">134</a>
   <a href="#appendix-A.1">A.1</a>        Client ..............................................  <a href="#page-134">134</a>
   <a href="#appendix-A.2">A.2</a>        Server ..............................................  <a href="#page-135">135</a>
   <a href="#appendix-A.3">A.3</a>        Header Processing ...................................  <a href="#page-135">135</a>
   <a href="#appendix-B">B</a>          Usage of the Session Description Protocol (SDP)......  <a href="#page-136">136</a>
   <a href="#appendix-B.1">B.1</a>        Configuring Media Streams ...........................  <a href="#page-136">136</a>
   <a href="#appendix-B.2">B.2</a>        Setting SDP Values for Unicast ......................  <a href="#page-138">138</a>
   <a href="#appendix-B.3">B.3</a>        Multicast Operation .................................  <a href="#page-139">139</a>
   <a href="#appendix-B.4">B.4</a>        Delayed Media Streams ...............................  <a href="#page-139">139</a>
   <a href="#appendix-B.5">B.5</a>        Putting Media Streams on Hold .......................  <a href="#page-139">139</a>
   <a href="#appendix-B.6">B.6</a>        Subject and SDP "s=" Line ...........................  <a href="#page-140">140</a>
   <a href="#appendix-B.7">B.7</a>        The SDP "o=" Line ...................................  <a href="#page-140">140</a>
   <a href="#appendix-C">C</a>          Summary of Augmented BNF ............................  <a href="#page-141">141</a>
   <a href="#appendix-C.1">C.1</a>        Basic Rules .........................................  <a href="#page-143">143</a>
   <a href="#appendix-D">D</a>          Using SRV DNS Records ...............................  <a href="#page-146">146</a>
   <a href="#appendix-E">E</a>          IANA Considerations .................................  <a href="#page-148">148</a>
   <a href="#appendix-F">F</a>          Acknowledgments .....................................  <a href="#page-149">149</a>
   <a href="#appendix-G">G</a>          Authors' Addresses ..................................  <a href="#page-149">149</a>
   <a href="#appendix-H">H</a>          Bibliography ........................................  <a href="#page-150">150</a>
   <a href="#appendix-I">I</a>          Full Copyright Statement ............................  <a href="#page-153">153</a>





<span class="grey">Handley, et al.             Standards Track                     [Page 6]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-7" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a> Introduction</span>

<span class="h3"><a class="selflink" id="section-1.1" href="#section-1.1">1.1</a> Overview of SIP Functionality</span>

   The Session Initiation Protocol (SIP) is an application-layer control
   protocol that can establish, modify and terminate multimedia sessions
   or calls. These multimedia sessions include multimedia conferences,
   distance learning, Internet telephony and similar applications. SIP
   can invite both persons and "robots", such as a media storage
   service.  SIP can invite parties to both unicast and multicast
   sessions; the initiator does not necessarily have to be a member of
   the session to which it is inviting. Media and participants can be
   added to an existing session.

   SIP can be used to initiate sessions as well as invite members to
   sessions that have been advertised and established by other means.
   Sessions can be advertised using multicast protocols such as SAP,
   electronic mail, news groups, web pages or directories (LDAP), among
   others.

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. These facilities also enable personal mobility.
   In the parlance of telecommunications intelligent network services,
   this is defined as: "Personal mobility is the ability of end users to
   originate and receive calls and access subscribed telecommunication
   services on any terminal in any location, and the ability of the
   network to identify end users as they move. Personal mobility is
   based on the use of a unique personal identity (i.e., personal
   number)." [<a href="#ref-1" title="&quot;Emerging mobile and personal communication systems,&quot;">1</a>]. Personal mobility complements terminal mobility, i.e.,
   the ability to maintain communications when moving a single end
   system from one subnet to another.

   SIP supports five facets of establishing and terminating multimedia
   communications:

   User location: determination of the end system to be used for
        communication;

   User capabilities: determination of the media and media parameters to
        be used;

   User availability: determination of the willingness of the called
        party to engage in communications;

   Call setup: "ringing", establishment of call parameters at both
        called and calling party;




<span class="grey">Handley, et al.             Standards Track                     [Page 7]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-8" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Call handling: including transfer and termination of calls.

   SIP can also initiate multi-party calls using a multipoint control
   unit (MCU) or fully-meshed interconnection instead of multicast.
   Internet telephony gateways that connect Public Switched Telephone
   Network (PSTN) parties can also use SIP to set up calls between them.

   SIP is designed as part of the overall IETF multimedia data and
   control architecture currently incorporating protocols such as RSVP
   (<a href="./rfc2205">RFC 2205</a> [<a href="#ref-2" title="&quot;Resource ReSerVation protocol (RSVP) -- version 1 functional specification&quot;">2</a>]) for reserving network resources, the real-time
   transport protocol (RTP) (<a href="./rfc1889">RFC 1889</a> [<a href="#ref-3" title="&quot;RTP: a transport protocol for real-time applications&quot;">3</a>]) for transporting real-time
   data and providing QOS feedback, the real-time streaming protocol
   (RTSP) (<a href="./rfc2326">RFC 2326</a> [<a href="#ref-4" title="&quot;Real time streaming protocol (RTSP)&quot;">4</a>]) for controlling delivery of streaming media,
   the session announcement protocol (SAP) [<a href="#ref-5" title="&quot;SAP: Session announcement protocol,&quot;">5</a>] for advertising
   multimedia sessions via multicast and the session description
   protocol (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]) for describing multimedia sessions.
   However, the functionality and operation of SIP does not depend on
   any of these protocols.

   SIP can also be used in conjunction with other call setup and
   signaling protocols. In that mode, an end system uses SIP exchanges
   to determine the appropriate end system address and protocol from a
   given address that is protocol-independent. For example, SIP could be
   used to determine that the party can be reached via H.323 [<a href="#ref-7" title="&quot;Visual telephone systems and equipment for local area networks which provide a non- guaranteed quality of service,&quot;">7</a>], obtain
   the H.245 [<a href="#ref-8" title="&quot;Control protocol for multimedia communication,&quot;">8</a>] gateway and user address and then use H.225.0 [<a href="#ref-9" title="&quot;Media stream packetization and synchronization on non-guaranteed quality of service LANs,&quot;">9</a>] to
   establish the call.

   In another example, SIP might be used to determine that the callee is
   reachable via the PSTN and indicate the phone number to be called,
   possibly suggesting an Internet-to-PSTN gateway to be used.

   SIP does not offer conference control services such as floor control
   or voting and does not prescribe how a conference is to be managed,
   but SIP can be used to introduce conference control protocols. SIP
   does not allocate multicast addresses.

   SIP can invite users to sessions with and without resource
   reservation.  SIP does not reserve resources, but can convey to the
   invited system the information necessary to do this.

<span class="h3"><a class="selflink" id="section-1.2" href="#section-1.2">1.2</a> Terminology</span>

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in <a href="./rfc2119">RFC 2119</a> [<a href="#ref-10" title="&quot;Key words for use in RFCs to indicate requirement levels&quot;">10</a>]
   and indicate requirement levels for compliant SIP implementations.





<span class="grey">Handley, et al.             Standards Track                     [Page 8]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-9" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-1.3" href="#section-1.3">1.3</a> Definitions</span>

   This specification uses a number of terms to refer to the roles
   played by participants in SIP communications. The definitions of
   client, server and proxy are similar to those used by the Hypertext
   Transport Protocol (HTTP) (<a href="./rfc2068">RFC 2068</a> [<a href="#ref-11" title="&quot;Hypertext transfer protocol -- HTTP/1.1&quot;">11</a>]). The terms and generic
   syntax of URI and URL are defined in <a href="./rfc2396">RFC 2396</a> [<a href="#ref-12" title="&quot;Uniform resource identifiers (URI): generic syntax&quot;">12</a>]. The following
   terms have special significance for SIP.

   Call: A call consists of all participants in a conference invited by
        a common source. A SIP call is identified by a globally unique
        call-id (<a href="#section-6.12">Section 6.12</a>). Thus, if a user is, for example, invited
        to the same multicast session by several people, each of these
        invitations will be a unique call. A point-to-point Internet
        telephony conversation maps into a single SIP call. In a
        multiparty conference unit (MCU) based call-in conference, each
        participant uses a separate call to invite himself to the MCU.

   Call leg: A call leg is identified by the combination of Call-ID, To
        and From.

   Client: An application program that sends SIP requests. Clients may
        or may not interact directly with a human user.  User agents and
        proxies contain clients (and servers).

   Conference: A multimedia session (see below), identified by a common
        session description. A conference can have zero or more members
        and includes the cases of a multicast conference, a full-mesh
        conference and a two-party "telephone call", as well as
        combinations of these.  Any number of calls can be used to
        create a conference.

   Downstream: Requests sent in the direction from the caller to the
        callee (i.e., user agent client to user agent server).

   Final response: A response that terminates a SIP transaction, as
        opposed to a provisional response that does not. All 2xx, 3xx,
        4xx, 5xx and 6xx responses are final.

   Initiator, calling party, caller: The party initiating a conference
        invitation. Note that the calling party does not have to be the
        same as the one creating the conference.

   Invitation: A request sent to a user (or service) requesting
        participation in a session. A successful SIP invitation consists
        of two transactions: an INVITE request followed by an ACK
        request.




<span class="grey">Handley, et al.             Standards Track                     [Page 9]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-10" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Invitee, invited user, called party, callee: The person or service
        that the calling party is trying to invite to a conference.

   Isomorphic request or response: Two requests or responses are defined
        to be isomorphic for the purposes of this document if they have
        the same values for the Call-ID, To, From and CSeq header
        fields. In addition, isomorphic requests have to have the same
        Request-URI.

   Location server: See location service.

   Location service: A location service is used by a SIP redirect or
        proxy server to obtain information about a callee's possible
        location(s). Location services are offered by location servers.
        Location servers MAY be co-located with a SIP server, but the
        manner in which a SIP server requests location services is
        beyond the scope of this document.

   Parallel search: In a parallel search, a proxy issues several
        requests to possible user locations upon receiving an incoming
        request.  Rather than issuing one request and then waiting for
        the final response before issuing the next request as in a
        sequential search , a parallel search issues requests without
        waiting for the result of previous requests.

   Provisional response: A response used by the server to indicate
        progress, but that does not terminate a SIP transaction. 1xx
        responses are provisional, other responses are considered final.

   Proxy, proxy server: An intermediary program that acts as both a
        server and a client for the purpose of making requests on behalf
        of other clients. Requests are serviced internally or by passing
        them on, possibly after translation, to other servers. A proxy
        interprets, and, if necessary, rewrites a request message before
        forwarding it.

   Redirect server: A redirect server is a server that accepts a SIP
        request, maps the address into zero or more new addresses and
        returns these addresses to the client. Unlike a proxy server ,
        it does not initiate its own SIP request. Unlike a user agent
        server , it does not accept calls.

   Registrar: A registrar is a server that accepts REGISTER requests. A
        registrar is typically co-located with a proxy or redirect
        server and MAY offer location services.






<span class="grey">Handley, et al.             Standards Track                    [Page 10]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-11" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Ringback: Ringback is the signaling tone produced by the calling
        client's application indicating that a called party is being
        alerted (ringing).

   Server: A server is an application program that accepts requests in
        order to service requests and sends back responses to those
        requests.  Servers are either proxy, redirect or user agent
        servers or registrars.

   Session: From the SDP specification: "A multimedia session is a set
        of multimedia senders and receivers and the data streams flowing
        from senders to receivers. A multimedia conference is an example
        of a multimedia session." (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]) (A session as defined
        for SDP can comprise one or more RTP sessions.) As defined, a
        callee can be invited several times, by different calls, to the
        same session. If SDP is used, a session is defined by the
        concatenation of the user name , session id , network type ,
        address type and address elements in the origin field.

   (SIP) transaction: A SIP transaction occurs between a client and a
        server and comprises all messages from the first request sent
        from the client to the server up to a final (non-1xx) response
        sent from the server to the client. A transaction is identified
        by the CSeq sequence number (<a href="#section-6.17">Section 6.17</a>) within a single call
        leg.  The ACK request has the same CSeq number as the
        corresponding INVITE request, but comprises a transaction of its
        own.

   Upstream: Responses sent in the direction from the user agent server
        to the user agent client.

   URL-encoded: A character string encoded according to <a href="./rfc1738#section-2.2">RFC 1738,
        Section&nbsp;2.2</a> [<a href="#ref-13" title="&quot;Uniform resource locators (URL)&quot;">13</a>].

   User agent client (UAC), calling user agent: A user agent client is a
        client application that initiates the SIP request.

   User agent server (UAS), called user agent: A user agent server is a
        server application that contacts the user when a SIP request is
        received and that returns a response on behalf of the user. The
        response accepts, rejects or redirects the request.

   User agent (UA): An application which contains both a user agent
        client and user agent server.

   An application program MAY be capable of acting both as a client and
   a server. For example, a typical multimedia conference control
   application would act as a user agent client to initiate calls or to



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-12" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   invite others to conferences and as a user agent server to accept
   invitations. The properties of the different SIP server types are
   summarized in Table 1.


    property                   redirect  proxy   user agent  registrar
                                server   server    server
    __________________________________________________________________
    also acts as a SIP client     no      yes        no         no
    returns 1xx status           yes      yes       yes         yes
    returns 2xx status            no      yes       yes         yes
    returns 3xx status           yes      yes       yes         yes
    returns 4xx status           yes      yes       yes         yes
    returns 5xx status           yes      yes       yes         yes
    returns 6xx status            no      yes       yes         yes
    inserts Via header            no      yes        no         no
    accepts ACK                  yes      yes       yes         no


   Table 1: Properties of the different SIP server types


<span class="h3"><a class="selflink" id="section-1.4" href="#section-1.4">1.4</a> Overview of SIP Operation</span>

   This section explains the basic protocol functionality and operation.
   Callers and callees are identified by SIP addresses, described in
   <a href="#section-1.4.1">Section 1.4.1</a>. When making a SIP call, a caller first locates the
   appropriate server (<a href="#section-1.4.2">Section 1.4.2</a>) and then sends a SIP request
   (<a href="#section-1.4.3">Section 1.4.3</a>). The most common SIP operation is the invitation
   (<a href="#section-1.4.4">Section 1.4.4</a>). Instead of directly reaching the intended callee, a
   SIP request may be redirected or may trigger a chain of new SIP
   requests by proxies (<a href="#section-1.4.5">Section 1.4.5</a>). Users can register their
   location(s) with SIP servers (<a href="#section-4.2.6">Section 4.2.6</a>).

<span class="h4"><a class="selflink" id="section-1.4.1" href="#section-1.4.1">1.4.1</a> SIP Addressing</span>

   The "objects" addressed by SIP are users at hosts, identified by a
   SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
   i.e., user@host.  The user part is a user name or a telephone number.
   The host part is either a domain name or a numeric network address.
   See <a href="#section-2">section 2</a> for a detailed discussion of SIP URL's.

   A user's SIP address can be obtained out-of-band, can be learned via
   existing media agents, can be included in some mailers' message
   headers, or can be recorded during previous invitation interactions.
   In many cases, a user's SIP URL can be guessed from their email
   address.




<span class="grey">Handley, et al.             Standards Track                    [Page 12]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-13" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   A SIP URL address can designate an individual (possibly located at
   one of several end systems), the first available person from a group
   of individuals or a whole group. The form of the address, for
   example, sip:sales@example.com , is not sufficient, in general, to
   determine the intent of the caller.

   If a user or service chooses to be reachable at an address that is
   guessable from the person's name and organizational affiliation, the
   traditional method of ensuring privacy by having an unlisted "phone"
   number is compromised. However, unlike traditional telephony, SIP
   offers authentication and access control mechanisms and can avail
   itself of lower-layer security mechanisms, so that client software
   can reject unauthorized or undesired call attempts.

<span class="h4"><a class="selflink" id="section-1.4.2" href="#section-1.4.2">1.4.2</a> Locating a SIP Server</span>

   When a client wishes to send a request, the client either sends it to
   a locally configured SIP proxy server (as in HTTP), independent of
   the Request-URI, or sends it to the IP address and port corresponding
   to the Request-URI.

   For the latter case, the client must determine the protocol, port and
   IP address of a server to which to send the request. A client SHOULD
   follow the steps below to obtain this information, but MAY follow the
   alternative, optional procedure defined in <a href="#appendix-D">Appendix D</a>. At each step,
   unless stated otherwise, the client SHOULD try to contact a server at
   the port number listed in the Request-URI. If no port number is
   present in the Request-URI, the client uses port 5060. If the
   Request-URI specifies a protocol (TCP or UDP), the client contacts
   the server using that protocol. If no protocol is specified, the
   client tries UDP (if UDP is supported). If the attempt fails, or if
   the client doesn't support UDP but supports TCP, it then tries TCP.

   A client SHOULD be able to interpret explicit network notifications
   (such as ICMP messages) which indicate that a server is not
   reachable, rather than relying solely on timeouts. (For socket-based
   programs: For TCP, connect() returns ECONNREFUSED if the client could
   not connect to a server at that address. For UDP, the socket needs to
   be bound to the destination address using connect() rather than
   sendto() or similar so that a second write() fails with ECONNREFUSED
   if there is no server listening) If the client finds the server is
   not reachable at a particular address, it SHOULD behave as if it had
   received a 400-class error response to that request.

   The client tries to find one or more addresses for the SIP server by
   querying DNS. The procedure is as follows:





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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-14" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        1.   If the host portion of the Request-URI is an IP address,
             the client contacts the server at the given address.
             Otherwise, the client proceeds to the next step.

        2.   The client queries the DNS server for address records for
             the host portion of the Request-URI. If the DNS server
             returns no address records, the client stops, as it has
             been unable to locate a server. By address record, we mean
             A RR's, AAAA RR's, or other similar address records, chosen
             according to the client's network protocol capabilities.


        There are no mandatory rules on how to select a host name
        for a SIP server. Users are encouraged to name their SIP
        servers using the sip.domainname (i.e., sip.example.com)
        convention, as specified in <a href="./rfc2219">RFC 2219</a> [<a href="#ref-16" title="&quot;Use of DNS aliases for network services&quot;">16</a>]. Users may only
        know an email address instead of a full SIP URL for a
        callee, however. In that case, implementations may be able
        to increase the likelihood of reaching a SIP server for
        that domain by constructing a SIP URL from that email
        address by prefixing the host name with "sip.". In the
        future, this mechanism is likely to become unnecessary as
        better DNS techniques, such as the one in <a href="#appendix-D">Appendix D</a>,
        become widely available.

   A client MAY cache a successful DNS query result. A successful query
   is one which contained records in the answer, and a server was
   contacted at one of the addresses from the answer. When the client
   wishes to send a request to the same host, it MUST start the search
   as if it had just received this answer from the name server. The
   client MUST follow the procedures in <a href="./rfc1035">RFC1035</a> [<a href="#ref-15" title="&quot;Domain names - implementation and specification&quot;">15</a>] regarding DNS cache
   invalidation when the DNS time-to-live expires.

<span class="h4"><a class="selflink" id="section-1.4.3" href="#section-1.4.3">1.4.3</a> SIP Transaction</span>

   Once the host part has been resolved to a SIP server, the client
   sends one or more SIP requests to that server and receives one or
   more responses from the server. A request (and its retransmissions)
   together with the responses triggered by that request make up a SIP
   transaction.  All responses to a request contain the same values in
   the Call-ID, CSeq, To, and From fields (with the possible addition of
   a tag in the To field (<a href="#section-6.37">section 6.37</a>)). This allows responses to be
   matched with requests. The ACK request following an INVITE is not
   part of the transaction since it may traverse a different set of
   hosts.






<span class="grey">Handley, et al.             Standards Track                    [Page 14]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-15" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   If TCP is used, request and responses within a single SIP transaction
   are carried over the same TCP connection (see <a href="#section-10">Section 10</a>). Several
   SIP requests from the same client to the same server MAY use the same
   TCP connection or MAY use a new connection for each request.

   If the client sent the request via unicast UDP, the response is sent
   to the address contained in the next Via header field (<a href="#section-6.40">Section 6.40</a>)
   of the response. If the request is sent via multicast UDP, the
   response is directed to the same multicast address and destination
   port. For UDP, reliability is achieved using retransmission (<a href="#section-10">Section</a>
   <a href="#section-10">10</a>).

   The SIP message format and operation is independent of the transport
   protocol.

<span class="h4"><a class="selflink" id="section-1.4.4" href="#section-1.4.4">1.4.4</a> SIP Invitation</span>

   A successful SIP invitation consists of two requests, INVITE followed
   by ACK. The INVITE (<a href="#section-4.2.1">Section 4.2.1</a>) request asks the callee to join a
   particular conference or establish a two-party conversation. After
   the callee has agreed to participate in the call, the caller confirms
   that it has received that response by sending an ACK (<a href="#section-4.2.2">Section 4.2.2</a>)
   request. If the caller no longer wants to participate in the call, it
   sends a BYE request instead of an ACK.

   The INVITE request typically contains a session description, for
   example written in SDP (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]) format, that provides the
   called party with enough information to join the session. For
   multicast sessions, the session description enumerates the media
   types and formats that are allowed to be distributed to that session.
   For a unicast session, the session description enumerates the media
   types and formats that the caller is willing to use and where it
   wishes the media data to be sent. In either case, if the callee
   wishes to accept the call, it responds to the invitation by returning
   a similar description listing the media it wishes to use. For a
   multicast session, the callee SHOULD only return a session
   description if it is unable to receive the media indicated in the
   caller's description or wants to receive data via unicast.

   The protocol exchanges for the INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. (Note that the
   messages shown in the figures have been abbreviated slightly.) In
   Fig. 1, the proxy server accepts the INVITE request (step 1),
   contacts the location service with all or parts of the address (step
   2) and obtains a more precise location (step 3). The proxy server
   then issues a SIP INVITE request to the address(es) returned by the
   location service (step 4). The user agent server alerts the user
   (step 5) and returns a success indication to the proxy server (step



<span class="grey">Handley, et al.             Standards Track                    [Page 15]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-16" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   6). The proxy server then returns the success result to the original
   caller (step 7). The receipt of this message is confirmed by the
   caller using an ACK request, which is forwarded to the callee (steps
   8 and 9). Note that an ACK can also be sent directly to the callee,
   bypassing the proxy. All requests and responses have the same Call-
   ID.





                                         +....... cs.columbia.edu .......+
                                         :                               :
                                         : (~~~~~~~~~~)                  :
                                         : ( location )                  :
                                         : ( service  )                  :
                                         : (~~~~~~~~~~)                  :
                                         :     ^    |                    :
                                         :     | hgs@lab                 :
                                         :    2|   3|                    :
                                         :     |    |                    :
                                         : henning  |                    :
+.. cs.tu-berlin.de ..+ 1: INVITE        :     |    |                    :
:                     :    henning@cs.col:     |   \/ 4: INVITE  5: ring :
: cz@cs.tu-berlin.de ========================&gt;(~~~~~~)=========&gt;(~~~~~~) :
:                    &lt;........................(      )&lt;.........(      ) :
:                     : 7: 200 OK        :    (      )6: 200 OK (      ) :
:                     :                  :    ( work )          ( lab  ) :
:                     : 8: ACK           :    (      )9: ACK    (      ) :
:                    ========================&gt;(~~~~~~)=========&gt;(~~~~~~) :
+.....................+                  +...............................+

  ====&gt; SIP request
  ....&gt; SIP response

   ^
   |    non-SIP protocols
   |


   Figure 1: Example of SIP proxy server



   The redirect server shown in Fig. 2 accepts the INVITE request (step
   1), contacts the location service as before (steps 2 and 3) and,
   instead of contacting the newly found address itself, returns the
   address to the caller (step 4), which is then acknowledged via an ACK



<span class="grey">Handley, et al.             Standards Track                    [Page 16]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-17" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   request (step 5). The caller issues a new request, with the same
   call-ID but a higher CSeq, to the address returned by the first
   server (step 6). In the example, the call succeeds (step 7). The
   caller and callee complete the handshake with an ACK (step 8).


   The next section discusses what happens if the location service
   returns more than one possible alternative.

<span class="h4"><a class="selflink" id="section-1.4.5" href="#section-1.4.5">1.4.5</a> Locating a User</span>

   A callee may move between a number of different end systems over
   time.  These locations can be dynamically registered with the SIP
   server (Sections <a href="#section-1.4.7">1.4.7</a>, <a href="#section-4.2.6">4.2.6</a>). A location server MAY also use one or
   more other protocols, such as finger (<a href="./rfc1288">RFC 1288</a> [<a href="#ref-17" title="&quot;The finger user information protocol&quot;">17</a>]), rwhois (<a href="./rfc2167">RFC</a>
   <a href="./rfc2167">2167</a> [<a href="#ref-18" title="&quot;Referral whois (rwhois) protocol V1.5&quot;">18</a>]), LDAP (<a href="./rfc1777">RFC 1777</a> [<a href="#ref-19" title="&quot;Lightweight directory access protocol&quot;">19</a>]), multicast-based protocols [<a href="#ref-20" title="&quot;A multicast user directory service for synchronous rendezvous,&quot;">20</a>] or
   operating-system dependent mechanisms to actively determine the end
   system where a user might be reachable. A location server MAY return
   several locations because the user is logged in at several hosts
   simultaneously or because the location server has (temporarily)
   inaccurate information. The SIP server combines the results to yield
   a list of a zero or more locations.

   The action taken on receiving a list of locations varies with the
   type of SIP server. A SIP redirect server returns the list to the
   client as Contact headers (<a href="#section-6.13">Section 6.13</a>). A SIP proxy server can
   sequentially or in parallel try the addresses until the call is
   successful (2xx response) or the callee has declined the call (6xx
   response). With sequential attempts, a proxy server can implement an
   "anycast" service.

   If a proxy server forwards a SIP request, it MUST add itself to the
   beginning of the list of forwarders noted in the Via (<a href="#section-6.40">Section 6.40</a>)
   headers. The Via trace ensures that replies can take the same path
   back, ensuring correct operation through compliant firewalls and
   avoiding request loops. On the response path, each host MUST remove
   its Via, so that routing internal information is hidden from the
   callee and outside networks. A proxy server MUST check that it does
   not generate a request to a host listed in the Via sent-by, via-
   received or via-maddr parameters (<a href="#section-6.40">Section 6.40</a>). (Note: If a host has
   several names or network addresses, this does not always work.  Thus,
   each host also checks if it is part of the Via list.)

   A SIP invitation may traverse more than one SIP proxy server. If one
   of these "forks" the request, i.e., issues more than one request in
   response to receiving the invitation request, it is possible that a
   client is reached, independently, by more than one copy of the




<span class="grey">Handley, et al.             Standards Track                    [Page 17]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-18" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   invitation request. Each of these copies bears the same Call-ID. The
   user agent MUST return the same status response returned in the first
   response. Duplicate requests are not an error.

<span class="h4"><a class="selflink" id="section-1.4.6" href="#section-1.4.6">1.4.6</a> Changing an Existing Session</span>

   In some circumstances, it is desirable to change the parameters of an
   existing session. This is done by re-issuing the INVITE, using the
   same Call-ID, but a new or different body or header fields to convey
   the new information. This re INVITE MUST have a higher CSeq than any
   previous request from the client to the server.

   For example, two parties may have been conversing and then want to
   add a third party, switching to multicast for efficiency.  One of the
   participants invites the third party with the new multicast address
   and simultaneously sends an INVITE to the second party, with the new
   multicast session description, but with the old call identifier.

<span class="h4"><a class="selflink" id="section-1.4.7" href="#section-1.4.7">1.4.7</a> Registration Services</span>

   The REGISTER request allows a client to let a proxy or redirect
   server know at which address(es) it can be reached. A client MAY also
   use it to install call handling features at the server.

<span class="h3"><a class="selflink" id="section-1.5" href="#section-1.5">1.5</a> Protocol Properties</span>

<span class="h4"><a class="selflink" id="section-1.5.1" href="#section-1.5.1">1.5.1</a> Minimal State</span>

   A single conference session or call involves one or more SIP
   request-response transactions. Proxy servers do not have to keep
   state for a particular call, however, they MAY maintain state for a
   single SIP transaction, as discussed in <a href="#section-12">Section 12</a>. For efficiency, a
   server MAY cache the results of location service requests.

<span class="h4"><a class="selflink" id="section-1.5.2" href="#section-1.5.2">1.5.2</a> Lower-Layer-Protocol Neutral</span>

   SIP makes minimal assumptions about the underlying transport and
   network-layer protocols. The lower-layer can provide either a packet
   or a byte stream service, with reliable or unreliable service.

   In an Internet context, SIP is able to utilize both UDP and TCP as
   transport protocols, among others. UDP allows the application to more
   carefully control the timing of messages and their retransmission, to
   perform parallel searches without requiring TCP connection state for
   each outstanding request, and to use multicast. Routers can more
   readily snoop SIP UDP packets. TCP allows easier passage through
   existing firewalls.




<span class="grey">Handley, et al.             Standards Track                    [Page 18]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-19" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>






                                         +....... cs.columbia.edu .......+
                                         :                               :
                                         : (~~~~~~~~~~)                  :
                                         : ( location )                  :
                                         : ( service  )                  :
                                         : (~~~~~~~~~~)                  :
                                         :    ^   |                      :
                                         :    | hgs@lab                  :
                                         :   2|  3|                      :
                                         :    |   |                      :
                                         : henning|                      :
+.. cs.tu-berlin.de ..+ 1: INVITE        :    |   |                      :
:                     :    henning@cs.col:    |   \/                     :
: cz@cs.tu-berlin.de =======================&gt;(~~~~~~)                    :
:       | ^ |        &lt;.......................(      )                    :
:       | . |         : 4: 302 Moved     :   (      )                    :
:       | . |         :    hgs@lab       :   ( work )                    :
:       | . |         :                  :   (      )                    :
:       | . |         : 5: ACK           :   (      )                    :
:       | . |        =======================&gt;(~~~~~~)                    :
:       | . |         :                  :                               :
+.......|...|.........+                  :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . | 6: INVITE hgs@lab.cs.columbia.edu                 (~~~~~~) :
        | . ==================================================&gt; (      ) :
        | ..................................................... (      ) :
        |     7: 200 OK                  :                      ( lab  ) :
        |                                :                      (      ) :
        |     8: ACK                     :                      (      ) :
        ======================================================&gt; (~~~~~~) :
                                         +...............................+

  ====&gt; SIP request
  ....&gt; SIP response

    ^
    |   non-SIP protocols
    |




   Figure 2: Example of SIP redirect server

<span class="grey">Handley, et al.             Standards Track                    [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   When TCP is used, SIP can use one or more connections to attempt to
   contact a user or to modify parameters of an existing conference.
   Different SIP requests for the same SIP call MAY use different TCP
   connections or a single persistent connection, as appropriate.

   For concreteness, this document will only refer to Internet
   protocols.  However, SIP MAY also be used directly with protocols
   such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
   conventions are beyond the scope of this document. User agents SHOULD
   implement both UDP and TCP transport. Proxy, registrar, and redirect
   servers MUST implement both UDP and TCP transport.

<span class="h4"><a class="selflink" id="section-1.5.3" href="#section-1.5.3">1.5.3</a> Text-Based</span>

   SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
   allows easy implementation in languages such as Java, Tcl and Perl,
   allows easy debugging, and most importantly, makes SIP flexible and
   extensible. As SIP is used for initiating multimedia conferences
   rather than delivering media data, it is believed that the additional
   overhead of using a text-based protocol is not significant.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a> SIP Uniform Resource Locators</span>

   SIP URLs are used within SIP messages to indicate the originator
   (From), current destination (Request-URI) and final recipient (To) of
   a SIP request, and to specify redirection addresses (Contact). A SIP
   URL can also be embedded in web pages or other hyperlinks to indicate
   that a particular user or service can be called via SIP. When used as
   a hyperlink, the SIP URL indicates the use of the INVITE method.

   The SIP URL scheme is defined to allow setting SIP request-header
   fields and the SIP message-body.


        This corresponds to the use of mailto: URLs. It makes it
        possible, for example, to specify the subject, urgency or
        media types of calls initiated through a web page or as
        part of an email message.

   A SIP URL follows the guidelines of <a href="./rfc2396">RFC 2396</a> [<a href="#ref-12" title="&quot;Uniform resource identifiers (URI): generic syntax&quot;">12</a>] and has the syntax
   shown in Fig. 3. The syntax is described using Augmented Backus-Naur
   Form (See Section C). Note that reserved characters have to be
   escaped and that the "set of characters reserved within any given URI
   component is defined by that component. In general, a character is
   reserved if the semantics of the URI changes if the character is
   replaced with its escaped US-ASCII encoding" [<a href="#ref-12" title="&quot;Uniform resource identifiers (URI): generic syntax&quot;">12</a>].





<span class="grey">Handley, et al.             Standards Track                    [Page 20]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




  SIP-URL         = "sip:" [ userinfo "@" ] hostport
                    url-parameters [ headers ]
  userinfo        = user [ ":" password ]
  user            = *( unreserved | escaped
                  | "&amp;" | "=" | "+" | "$" | "," )
  password        = *( unreserved | escaped
                  | "&amp;" | "=" | "+" | "$" | "," )
  hostport        = host [ ":" port ]
  host            = hostname | IPv4address
  hostname        = *( domainlabel "." ) toplabel [ "." ]
  domainlabel     = alphanum | alphanum *( alphanum | "-" ) alphanum
  toplabel        = alpha | alpha *( alphanum | "-" ) alphanum
  IPv4address     = 1*digit "." 1*digit "." 1*digit "." 1*digit
  port            = *digit
  url-parameters  = *( ";" url-parameter )
  url-parameter   = transport-param | user-param | method-param
                  | ttl-param | maddr-param | other-param
  transport-param = "transport=" ( "udp" | "tcp" )
  ttl-param       = "ttl=" ttl
  ttl             = 1*3DIGIT       ; 0 to 255
  maddr-param     = "maddr=" host
  user-param      = "user=" ( "phone" | "ip" )
  method-param    = "method=" Method
  tag-param       = "tag=" UUID
  UUID            = 1*( hex | "-" )
  other-param     = ( token | ( token "=" ( token | quoted-string )))
  headers         = "?" header *( "&amp;" header )
  header          = hname "=" hvalue
  hname           = 1*uric
  hvalue          = *uric
  uric            = reserved | unreserved | escaped
  reserved        = ";" | "/" | "?" | ":" | "@" | "&amp;" | "=" | "+" |
                    "$" | ","
  digits          = 1*DIGIT


   Figure 3: SIP URL syntax



   The URI character classes referenced above are described in <a href="#appendix-C">Appendix</a>
   <a href="#appendix-C">C</a>.

   The components of the SIP URI have the following meanings.





<span class="grey">Handley, et al.             Standards Track                    [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




telephone-subscriber  = global-phone-number | local-phone-number
   global-phone-number   = "+" 1*phonedigit [isdn-subaddress]
                             [post-dial]
   local-phone-number    = 1*(phonedigit | dtmf-digit |
                             pause-character) [isdn-subaddress]
                             [post-dial]
   isdn-subaddress       = ";isub=" 1*phonedigit
   post-dial             = ";postd=" 1*(phonedigit | dtmf-digit
                         |  pause-character)
   phonedigit            = DIGIT | visual-separator
   visual-separator      = "-" | "."
   pause-character       = one-second-pause | wait-for-dial-tone
   one-second-pause      = "p"
   wait-for-dial-tone    = "w"
   dtmf-digit            = "*" | "#" | "A" | "B" | "C" | "D"

   Figure 4: SIP URL syntax; telephone subscriber

   user: If the host is an Internet telephony gateway, the user field
        MAY also encode a telephone number using the notation of
        telephone-subscriber (Fig. 4). The telephone number is a special
        case of a user name and cannot be distinguished by a BNF. Thus,
        a URL parameter, user, is added to distinguish telephone numbers
        from user names. The phone identifier is to be used when
        connecting to a telephony gateway. Even without this parameter,
        recipients of SIP URLs MAY interpret the pre-@ part as a phone
        number if local restrictions on the name space for user name
        allow it.

   password: The SIP scheme MAY use the format "user:password" in the
        userinfo field. The use of passwords in the userinfo is NOT
        RECOMMENDED, because the passing of authentication information
        in clear text (such as URIs) has proven to be a security risk in
        almost every case where it has been used.

   host: The mailto: URL and <a href="./rfc822">RFC 822</a> email addresses require that
        numeric host addresses ("host numbers") are enclosed in square
        brackets (presumably, since host names might be numeric), while
        host numbers without brackets are used for all other URLs. The
        SIP URL requires the latter form, without brackets.

   The issue of IPv6 literal addresses in URLs is being looked at
   elsewhere in the IETF. SIP implementers are advised to keep up to
   date on that activity.




<span class="grey">Handley, et al.             Standards Track                    [Page 22]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-23" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   port: The port number to send a request to. If not present, the
        procedures outlined in <a href="#section-1.4.2">Section 1.4.2</a> are used to determine the
        port number to send a request to.

   URL parameters: SIP URLs can define specific parameters of the
        request. URL parameters are added after the host component and
        are separated by semi-colons. The transport parameter determines
        the the transport mechanism (UDP or TCP). UDP is to be assumed
        when no explicit transport parameter is included. The maddr
        parameter provides the server address to be contacted for this
        user, overriding the address supplied in the host field.  This
        address is typically a multicast address, but could also be the
        address of a backup server. The ttl parameter determines the
        time-to-live value of the UDP multicast packet and MUST only be
        used if maddr is a multicast address and the transport protocol
        is UDP. The user parameter was described above. For example, to
        specify to call j.doe@big.com using multicast to 239.255.255.1
        with a ttl of 15, the following URL would be used:


     sip:j.doe@big.com;maddr=239.255.255.1;ttl=15



   The transport, maddr, and ttl parameters MUST NOT be used in the From
   and To header fields and the Request-URI; they are ignored if
   present.

   Headers: Headers of the SIP request can be defined with the "?"
        mechanism within a SIP URL. The special hname "body" indicates
        that the associated hvalue is the message-body of the SIP INVITE
        request. Headers MUST NOT be used in the From and To header
        fields and the Request-URI; they are ignored if present.  hname
        and hvalue are encodings of a SIP header name and value,
        respectively. All URL reserved characters in the header names
        and values MUST be escaped.

   Method: The method of the SIP request can be specified with the
        method parameter.  This parameter MUST NOT be used in the From
        and To header fields and the Request-URI; they are ignored if
        present.

   Table 2 summarizes where the components of the SIP URL can be used
   and what default values they assume if not present.


   Examples of SIP URLs are:




<span class="grey">Handley, et al.             Standards Track                    [Page 23]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-24" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>



                     default    Req.-URI  To  From  Contact  external
      user           --         x         x   x     x        x
      password       --         x         x         x        x
      host           mandatory  x         x   x     x        x
      port           5060       x         x   x     x        x
      user-param     ip         x         x   x     x        x
      method         INVITE                         x        x
      maddr-param    --                             x        x
      ttl-param      1                              x        x
      transp.-param  --                             x        x
      headers        --                             x        x


   Table 2: Use and default values of URL components  for  SIP  headers,
   Request-URI and references

     sip:j.doe@big.com
     sip:j.doe:secret@big.com;transport=tcp
     sip:j.doe@big.com?subject=project
     sip:+1-212-555-1212:1234@gateway.com;user=phone
     sip:1212@gateway.com
     sip:alice@10.1.2.3
     sip:alice@example.com
     sip:alice%40example.com@gateway.com
     sip:alice@registrar.com;method=REGISTER



   Within a SIP message, URLs are used to indicate the source and
   intended destination of a request, redirection addresses and the
   current destination of a request. Normally all these fields will
   contain SIP URLs.

   SIP URLs are case-insensitive, so that for example the two URLs
   sip:j.doe@example.com and SIP:J.Doe@Example.com are equivalent.  All
   URL parameters are included when comparing SIP URLs for equality.

   SIP header fields MAY contain non-SIP URLs. As an example, if a call
   from a telephone is relayed to the Internet via SIP, the SIP From
   header field might contain a phone URL.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a> SIP Message Overview</span>

   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (<a href="./rfc2279">RFC 2279</a> [<a href="#ref-21" title="&quot;UTF-8, a transformation format of ISO 10646&quot;">21</a>]). Senders MUST terminate lines with a
   CRLF, but receivers MUST also interpret CR and LF by themselves as
   line terminators.



<span class="grey">Handley, et al.             Standards Track                    [Page 24]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-25" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Except for the above difference in character sets, much of the
   message syntax is and header fields are identical to HTTP/1.1; rather
   than repeating the syntax and semantics here we use [HX.Y] to refer
   to Section X.Y of the current HTTP/1.1 specification (<a href="./rfc2068">RFC 2068</a> [<a href="#ref-11" title="&quot;Hypertext transfer protocol -- HTTP/1.1&quot;">11</a>]).
   In addition, we describe SIP in both prose and an augmented Backus-
   Naur form (ABNF). See section C for an overview of ABNF.

   Note, however, that SIP is not an extension of HTTP.

   Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
   transactions can be carried in a single TCP connection or UDP
   datagram. UDP datagrams, including all headers, SHOULD NOT be larger
   than the path maximum transmission unit (MTU) if the MTU is known, or
   1500 bytes if the MTU is unknown.


        The 1500 bytes accommodates encapsulation within the
        "typical" ethernet MTU without IP fragmentation. Recent
        studies [<a href="#ref-22" title="TCP/IP illustrated: the protocols">22</a>] indicate that an MTU of 1500 bytes is a
        reasonable assumption. The next lower common MTU values are
        1006 bytes for SLIP and 296 for low-delay PPP (<a href="./rfc1191">RFC 1191</a>
        [<a href="#ref-23" title="&quot;Path MTU discovery&quot;">23</a>]). Thus, another reasonable value would be a message
        size of 950 bytes, to accommodate packet headers within the
        SLIP MTU without fragmentation.

   A SIP message is either a request from a client to a server, or a
   response from a server to a client.



        SIP-message  =  Request | Response


   Both Request (<a href="#section-4">section 4</a>) and Response (<a href="#section-5">section 5</a>) messages use the
   generic-message format of <a href="./rfc822">RFC 822</a> [<a href="#ref-24" title="&quot;Standard for the format of ARPA internet text messages&quot;">24</a>] for transferring entities (the
   body of the message). Both types of messages consist of a start-line,
   one or more header fields (also known as "headers"), an empty line
   (i.e., a line with nothing preceding the carriage-return line-feed
   (CRLF)) indicating the end of the header fields, and an optional
   message-body. To avoid confusion with similar-named headers in HTTP,
   we refer to the headers describing the message body as entity
   headers. These components are described in detail in the upcoming
   sections.



        generic-message  =  start-line
                            *message-header



<span class="grey">Handley, et al.             Standards Track                    [Page 25]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-26" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


                            CRLF
                            [ message-body ]

        start-line       =  Request-Line |     ;<a href="#section-4.1">Section 4.1</a>
                            Status-Line        ;<a href="#section-5.1">Section 5.1</a>




        message-header  =  ( general-header
                           | request-header
                           | response-header
                           | entity-header )



   In the interest of robustness, any leading empty line(s) MUST be
   ignored. In other words, if the Request or Response message begins
   with one or more CRLF, CR, or LFs, these characters MUST be ignored.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a> Request</span>

   The Request message format is shown below:



        Request  =  Request-Line       ;  <a href="#section-4.1">Section 4.1</a>
                    *( general-header
                    | request-header
                    | entity-header )
                    CRLF
                    [ message-body ]   ;  <a href="#section-8">Section 8</a>


<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a> Request-Line</span>

   The Request-Line begins with a method token, followed by the
   Request-URI and the protocol version, and ending with CRLF. The
   elements are separated by SP characters.  No CR or LF are allowed
   except in the final CRLF sequence.



        Request-Line  =  Method SP Request-URI SP SIP-Version CRLF







<span class="grey">Handley, et al.             Standards Track                    [Page 26]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-27" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




        general-header   =  Accept               ; <a href="#section-6.7">Section 6.7</a>
                         |  Accept-Encoding      ; <a href="#section-6.8">Section 6.8</a>
                         |  Accept-Language      ; <a href="#section-6.9">Section 6.9</a>
                         |  Call-ID              ; <a href="#section-6.12">Section 6.12</a>
                         |  Contact              ; <a href="#section-6.13">Section 6.13</a>
                         |  CSeq                 ; <a href="#section-6.17">Section 6.17</a>
                         |  Date                 ; <a href="#section-6.18">Section 6.18</a>
                         |  Encryption           ; <a href="#section-6.19">Section 6.19</a>
                         |  Expires              ; <a href="#section-6.20">Section 6.20</a>
                         |  From                 ; <a href="#section-6.21">Section 6.21</a>
                         |  Record-Route         ; <a href="#section-6.29">Section 6.29</a>
                         |  Timestamp            ; <a href="#section-6.36">Section 6.36</a>
                         |  To                   ; <a href="#section-6.37">Section 6.37</a>
                         |  Via                  ; <a href="#section-6.40">Section 6.40</a>
        entity-header    =  Content-Encoding     ; <a href="#section-6.14">Section 6.14</a>
                         |  Content-Length       ; <a href="#section-6.15">Section 6.15</a>
                         |  Content-Type         ; <a href="#section-6.16">Section 6.16</a>
        request-header   =  Authorization        ; <a href="#section-6.11">Section 6.11</a>
                         |  Contact              ; <a href="#section-6.13">Section 6.13</a>
                         |  Hide                 ; <a href="#section-6.22">Section 6.22</a>
                         |  Max-Forwards         ; <a href="#section-6.23">Section 6.23</a>
                         |  Organization         ; <a href="#section-6.24">Section 6.24</a>
                         |  Priority             ; <a href="#section-6.25">Section 6.25</a>
                         |  Proxy-Authorization  ; <a href="#section-6.27">Section 6.27</a>
                         |  Proxy-Require        ; <a href="#section-6.28">Section 6.28</a>
                         |  Route                ; <a href="#section-6.33">Section 6.33</a>
                         |  Require              ; <a href="#section-6.30">Section 6.30</a>
                         |  Response-Key         ; <a href="#section-6.31">Section 6.31</a>
                         |  Subject              ; <a href="#section-6.35">Section 6.35</a>
                         |  User-Agent           ; <a href="#section-6.39">Section 6.39</a>
        response-header  =  Allow                ; <a href="#section-6.10">Section 6.10</a>
                         |  Proxy-Authenticate   ; <a href="#section-6.26">Section 6.26</a>
                         |  Retry-After          ; <a href="#section-6.32">Section 6.32</a>
                         |  Server               ; <a href="#section-6.34">Section 6.34</a>
                         |  Unsupported          ; <a href="#section-6.38">Section 6.38</a>
                         |  Warning              ; <a href="#section-6.41">Section 6.41</a>
                         |  WWW-Authenticate     ; <a href="#section-6.42">Section 6.42</a>


   Table 3: SIP headers

<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a> Methods</span>

   The methods are defined below. Methods that are not supported by a
   proxy or redirect server are treated by that server as if they were
   an OPTIONS method and forwarded accordingly. Methods that are not



<span class="grey">Handley, et al.             Standards Track                    [Page 27]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-28" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   supported by a user agent server or registrar cause a 501 (Not
   Implemented) response to be returned (<a href="#section-7">Section 7</a>). As in HTTP, the
   Method token is case-sensitive.



        Method  =  "INVITE" | "ACK" | "OPTIONS" | "BYE"
                   | "CANCEL" | "REGISTER"


<span class="h4"><a class="selflink" id="section-4.2.1" href="#section-4.2.1">4.2.1</a> INVITE</span>

   The INVITE method indicates that the user or service is being invited
   to participate in a session. The message body contains a description
   of the session to which the callee is being invited. For two-party
   calls, the caller indicates the type of media it is able to receive
   and possibly the media it is willing to send as well as their
   parameters such as network destination. A success response MUST
   indicate in its message body which media the callee wishes to receive
   and MAY indicate the media the callee is going to send.


        Not all session description formats have the ability to
        indicate sending media.

   A server MAY automatically respond to an invitation for a conference
   the user is already participating in, identified either by the SIP
   Call-ID or a globally unique identifier within the session
   description, with a 200 (OK) response.

   If a user agent receives an INVITE request for an existing call leg
   with a higher CSeq sequence number than any previous INVITE for the
   same Call-ID, it MUST check any version identifiers in the session
   description or, if there are no version identifiers, the content of
   the session description to see if it has changed. It MUST also
   inspect any other header fields for changes. If there is a change,
   the user agent MUST update any internal state or information
   generated as a result of that header. If the session description has
   changed, the user agent server MUST adjust the session parameters
   accordingly, possibly after asking the user for confirmation.
   (Versioning of the session description can be used to accommodate the
   capabilities of new arrivals to a conference, add or delete media or
   change from a unicast to a multicast conference.)

   This method MUST be supported by SIP proxy, redirect and user agent
   servers as well as clients.





<span class="grey">Handley, et al.             Standards Track                    [Page 28]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-29" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-4.2.2" href="#section-4.2.2">4.2.2</a> ACK</span>

   The ACK request confirms that the client has received a final
   response to an INVITE request. (ACK is used only with INVITE
   requests.) 2xx responses are acknowledged by client user agents, all
   other final responses by the first proxy or client user agent to
   receive the response. The Via is always initialized to the host that
   originates the ACK request, i.e., the client user agent after a 2xx
   response or the first proxy to receive a non-2xx final response. The
   ACK request is forwarded as the corresponding INVITE request, based
   on its Request-URI. See <a href="#section-10">Section 10</a> for details.

   The ACK request MAY contain a message body with the final session
   description to be used by the callee. If the ACK message body is
   empty, the callee uses the session description in the INVITE request.

   A proxy server receiving an ACK request after having sent a 3xx, 4xx,
   5xx, or 6xx response must make a determination about whether the ACK
   is for it, or for some user agent or proxy server further downstream.
   This determination is made by examining the tag in the To field. If
   the tag in the ACK To header field matches the tag in the To header
   field of the response, and the From, CSeq and Call-ID header fields
   in the response match those in the ACK, the ACK is meant for the
   proxy server. Otherwise, the ACK SHOULD be proxied downstream as any
   other request.


        It is possible for a user agent client or proxy server to
        receive multiple 3xx, 4xx, 5xx, and 6xx responses to a
        request along a single branch. This can happen under
        various error conditions, typically when a forking proxy
        transitions from stateful to stateless before receiving all
        responses. The various responses will all be identical,
        except for the tag in the To field, which is different for
        each one. It can therefore be used as a means to
        disambiguate them.

   This method MUST be supported by SIP proxy, redirect and user agent
   servers as well as clients.

<span class="h4"><a class="selflink" id="section-4.2.3" href="#section-4.2.3">4.2.3</a> OPTIONS</span>

   The server is being queried as to its capabilities. A server that
   believes it can contact the user, such as a user agent where the user
   is logged in and has been recently active, MAY respond to this
   request with a capability set. A called user agent MAY return a
   status reflecting how it would have responded to an invitation, e.g.,




<span class="grey">Handley, et al.             Standards Track                    [Page 29]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-30" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   600 (Busy). Such a server SHOULD return an Allow header field
   indicating the methods that it supports. Proxy and redirect servers
   simply forward the request without indicating their capabilities.

   This method MUST be supported by SIP proxy, redirect and user agent
   servers, registrars and clients.

<span class="h4"><a class="selflink" id="section-4.2.4" href="#section-4.2.4">4.2.4</a> BYE</span>

   The user agent client uses BYE to indicate to the server that it
   wishes to release the call. A BYE request is forwarded like an INVITE
   request and MAY be issued by either caller or callee. A party to a
   call SHOULD issue a BYE request before releasing a call ("hanging
   up"). A party receiving a BYE request MUST cease transmitting media
   streams specifically directed at the party issuing the BYE request.

   If the INVITE request contained a Contact header, the callee SHOULD
   send a BYE request to that address rather than the From address.

   This method MUST be supported by proxy servers and SHOULD be
   supported by redirect and user agent SIP servers.

<span class="h4"><a class="selflink" id="section-4.2.5" href="#section-4.2.5">4.2.5</a> CANCEL</span>

   The CANCEL request cancels a pending request with the same Call-ID,
   To, From and CSeq (sequence number only) header field values, but
   does not affect a completed request. (A request is considered
   completed if the server has returned a final status response.)

   A user agent client or proxy client MAY issue a CANCEL request at any
   time. A proxy, in particular, MAY choose to send a CANCEL to
   destinations that have not yet returned a final response after it has
   received a 2xx or 6xx response for one or more of the parallel-search
   requests. A proxy that receives a CANCEL request forwards the request
   to all destinations with pending requests.

   The Call-ID, To, the numeric part of CSeq and From headers in the
   CANCEL request are identical to those in the original request. This
   allows a CANCEL request to be matched with the request it cancels.
   However, to allow the client to distinguish responses to the CANCEL
   from those to the original request, the CSeq Method component is set
   to CANCEL. The Via header field is initialized to the proxy issuing
   the CANCEL request. (Thus, responses to this CANCEL request only
   reach the issuing proxy.)

   Once a user agent server has received a CANCEL, it MUST NOT issue a
   2xx response for the cancelled original request.




<span class="grey">Handley, et al.             Standards Track                    [Page 30]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-31" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   A redirect or user agent server receiving a CANCEL request responds
   with a status of 200 (OK) if the transaction exists and a status of
   481 (Transaction Does Not Exist) if not, but takes no further action.
   In particular, any existing call is unaffected.


        The BYE request cannot be used to cancel branches of a
        parallel search, since several branches may, through
        intermediate proxies, find the same user agent server and
        then terminate the call.  To terminate a call instead of
        just pending searches, the UAC must use BYE instead of or
        in addition to CANCEL. While CANCEL can terminate any
        pending request other than ACK or CANCEL, it is typically
        useful only for INVITE. 200 responses to INVITE and 200
        responses to CANCEL are distinguished by the method in the
        Cseq header field, so there is no ambiguity.

   This method MUST be supported by proxy servers and SHOULD be
   supported by all other SIP server types.

<span class="h4"><a class="selflink" id="section-4.2.6" href="#section-4.2.6">4.2.6</a> REGISTER</span>

   A client uses the REGISTER method to register the address listed in
   the To header field with a SIP server.

   A user agent MAY register with a local server on startup by sending a
   REGISTER request to the well-known "all SIP servers" multicast
   address "sip.mcast.net" (224.0.1.75). This request SHOULD be scoped
   to ensure it is not forwarded beyond the boundaries of the
   administrative system. This MAY be done with either TTL or
   administrative scopes [<a href="#ref-25" title="&quot;Administratively scoped IP multicast&quot;">25</a>], depending on what is implemented in the
   network. SIP user agents MAY listen to that address and use it to
   become aware of the location of other local users [<a href="#ref-20" title="&quot;A multicast user directory service for synchronous rendezvous,&quot;">20</a>]; however, they
   do not respond to the request.  A user agent MAY also be configured
   with the address of a registrar server to which it sends a REGISTER
   request upon startup.

   Requests are processed in the order received. Clients SHOULD avoid
   sending a new registration (as opposed to a retransmission) until
   they have received the response from the server for the previous one.


        Clients may register from different locations, by necessity
        using different Call-ID values. Thus, the CSeq value cannot
        be used to enforce ordering. Since registrations are
        additive, ordering is less of a problem than if each
        REGISTER request completely replaced all earlier ones.




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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The meaning of the REGISTER request-header fields is defined as
   follows. We define "address-of-record" as the SIP address that the
   registry knows the registrand, typically of the form "user@domain"
   rather than "user@host". In third-party registration, the entity
   issuing the request is different from the entity being registered.

   To: The To header field contains the address-of-record whose
        registration is to be created or updated.

   From: The From header field contains the address-of-record of the
        person responsible for the registration. For first-party
        registration, it is identical to the To header field value.

   Request-URI: The Request-URI names the destination of the
        registration request, i.e., the domain of the registrar. The
        user name MUST be empty. Generally, the domains in the Request-
        URI and the To header field have the same value; however, it is
        possible to register as a "visitor", while maintaining one's
        name. For example, a traveler sip:alice@acme.com (To) might
        register under the Request-URI sip:atlanta.hiayh.org , with the
        former as the To header field and the latter as the Request-URI.
        The REGISTER request is no longer forwarded once it has reached
        the server whose authoritative domain is the one listed in the
        Request-URI.

   Call-ID: All registrations from a client SHOULD use the same Call-ID
        header value, at least within the same reboot cycle.

   Cseq: Registrations with the same Call-ID MUST have increasing CSeq
        header values. However, the server does not reject out-of-order
        requests.

   Contact: The request MAY contain a Contact header field; future non-
        REGISTER requests for the URI given in the To header field
        SHOULD be directed to the address(es) given in the Contact
        header.

   If the request does not contain a Contact header, the registration
   remains unchanged.

        This is useful to obtain the current list of registrations
        in the response.  Registrations using SIP URIs that differ
        in one or more of host, port, transport-param or maddr-
        param (see Figure 3) from an existing registration are
        added to the list of registrations. Other URI types are
        compared according to the standard URI equivalency rules
        for the URI schema. If the URIs are equivalent to that of
        an existing registration, the new registration replaces the



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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        old one if it has a higher q value or, for the same value
        of q, if the ttl value is higher. All current registrations
        MUST share the same action value.  Registrations that have
        a different action than current registrations for the same
        user MUST be rejected with status of 409 (Conflict).

   A proxy server ignores the q parameter when processing non-REGISTER
   requests, while a redirect server simply returns that parameter in
   its Contact response header field.


        Having the proxy server interpret the q parameter is not
        sufficient to guide proxy behavior, as it is not clear, for
        example, how long it is supposed to wait between trying
        addresses.

   If the registration is changed while a user agent or proxy server
   processes an invitation, the new information SHOULD be used.


        This allows a service known as "directed pick-up". In the
        telephone network, directed pickup permits a user at a
        remote station who hears his own phone ringing to pick up
        at that station, dial an access code, and be connected to
        the calling user as if he had answered his own phone.

   A server MAY choose any duration for the registration lifetime.
   Registrations not refreshed after this amount of time SHOULD be
   silently discarded. Responses to a registration SHOULD include an
   Expires header (<a href="#section-6.20">Section 6.20</a>) or expires Contact parameters (<a href="#section-6.13">Section</a>
   <a href="#section-6.13">6.13</a>), indicating the time at which the server will drop the
   registration. If none is present, one hour is assumed. Clients MAY
   request a registration lifetime by indicating the time in an Expires
   header in the request. A server SHOULD NOT use a higher lifetime than
   the one requested, but MAY use a lower one. A single address (if
   host-independent) MAY be registered from several different clients.

   A client cancels an existing registration by sending a REGISTER
   request with an expiration time (Expires) of zero seconds for a
   particular Contact or the wildcard Contact designated by a "*" for
   all registrations. Registrations are matched based on the user, host,
   port and maddr parameters.

   The server SHOULD return the current list of registrations in the 200
   response as Contact header fields.

   It is particularly important that REGISTER requests are authenticated
   since they allow to redirect future requests (see <a href="#section-13.2">Section 13.2</a>).



<span class="grey">Handley, et al.             Standards Track                    [Page 33]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-34" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Beyond its use as a simple location service, this method is
        needed if there are several SIP servers on a single host.
        In that case, only one of the servers can use the default
        port number.


   Support of this method is RECOMMENDED.

<span class="h3"><a class="selflink" id="section-4.3" href="#section-4.3">4.3</a> Request-URI</span>

   The Request-URI is a SIP URL as described in <a href="#section-2">Section 2</a> or a general
   URI. It indicates the user or service to which this request is being
   addressed. Unlike the To field, the Request-URI MAY be re-written by
   proxies.

   When used as a Request-URI, a SIP-URL MUST NOT contain the
   transport-param, maddr-param, ttl-param, or headers elements. A
   server that receives a SIP-URL with these elements removes them
   before further processing.


        Typically, the UAC sets the Request-URI and To to the same
        SIP URL, presumed to remain unchanged over long time
        periods. However, if the UAC has cached a more direct path
        to the callee, e.g., from the Contact header field of a
        response to a previous request, the To would still contain
        the long-term, "public" address, while the Request-URI
        would be set to the cached address.

   Proxy and redirect servers MAY use the information in the Request-URI
   and request header fields to handle the request and possibly rewrite
   the Request-URI. For example, a request addressed to the generic
   address sip:sales@acme.com is proxied to the particular person, e.g.,
   sip:bob@ny.acme.com , with the To field remaining as
   sip:sales@acme.com.  At ny.acme.com , Bob then designates Alice as
   the temporary substitute.

   The host part of the Request-URI typically agrees with one of the
   host names of the receiving server. If it does not, the server SHOULD
   proxy the request to the address indicated or return a 404 (Not
   Found) response if it is unwilling or unable to do so. For example,
   the Request-URI and server host name can disagree in the case of a
   firewall proxy that handles outgoing calls. This mode of operation is
   similar to that of HTTP proxies.

   If a SIP server receives a request with a URI indicating a scheme
   other than SIP which that server does not understand, the server MUST
   return a 400 (Bad Request) response. It MUST do this even if the To



<span class="grey">Handley, et al.             Standards Track                    [Page 34]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-35" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   header field contains a scheme it does understand.  This is because
   proxies are responsible for processing the Request-URI; the To field
   is of end-to-end significance.

<span class="h4"><a class="selflink" id="section-4.3.1" href="#section-4.3.1">4.3.1</a> SIP Version</span>

   Both request and response messages include the version of SIP in use,
   and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced
   by SIP/2.0) regarding version ordering, compliance requirements, and
   upgrading of version numbers. To be compliant with this
   specification, applications sending SIP messages MUST include a SIP-
   Version of "SIP/2.0".

<span class="h3"><a class="selflink" id="section-4.4" href="#section-4.4">4.4</a> Option Tags</span>

   Option tags are unique identifiers used to designate new options in
   SIP.  These tags are used in Require (<a href="#section-6.30">Section 6.30</a>) and Unsupported
   (<a href="#section-6.38">Section 6.38</a>) fields.

   Syntax:


        option-tag  =  token


   See Section C for a definition of token. The creator of a new SIP
   option MUST either prefix the option with their reverse domain name
   or register the new option with the Internet Assigned Numbers
   Authority (IANA). For example, "com.foo.mynewfeature" is an apt name
   for a feature whose inventor can be reached at "foo.com".  Individual
   organizations are then responsible for ensuring that option names
   don't collide. Options registered with IANA have the prefix
   "org.iana.sip.", options described in RFCs have the prefix
   "org.ietf.rfc.N", where N is the RFC number. Option tags are case-
   insensitive.

<span class="h4"><a class="selflink" id="section-4.4.1" href="#section-4.4.1">4.4.1</a> Registering New Option Tags with IANA</span>

   When registering a new SIP option, the following information MUST be
   provided:

        o  Name and description of option. The name MAY be of any
          length, but SHOULD be no more than twenty characters long. The
          name MUST consist of alphanum (See Figure 3) characters only;







<span class="grey">Handley, et al.             Standards Track                    [Page 35]</span></pre>
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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        o  Indication of who has change control over the option (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium or a particular company or group of
          companies);

        o  A reference to a further description, if available, for
          example (in order of preference) an RFC, a published paper, a
          patent filing, a technical report, documented source code or a
          computer manual;

        o  Contact information (postal and email address);

   Registrations should be sent to iana@iana.org


        This procedure has been borrowed from RTSP [<a href="#ref-4" title="&quot;Real time streaming protocol (RTSP)&quot;">4</a>] and the RTP
        AVP [<a href="#ref-26" title="&quot;RTP profile for audio and video conferences with minimal control&quot;">26</a>].

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a> Response</span>

   After receiving and interpreting a request message, the recipient
   responds with a SIP response message. The response message format is
   shown below:



        Response  =  Status-Line        ;  <a href="#section-5.1">Section 5.1</a>
                     *( general-header
                     | response-header
                     | entity-header )
                     CRLF
                     [ message-body ]   ;  <a href="#section-8">Section 8</a>


   SIP's structure of responses is similar to [H6], but is defined
   explicitly here.

<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a> Status-Line</span>

   The first line of a Response message is the Status-Line, consisting
   of the protocol version (<a href="#section-4.3.1">Section 4.3.1</a>) followed by a numeric
   Status-Code and its associated textual phrase, with each element
   separated by SP characters. No CR or LF is allowed except in the
   final CRLF sequence.



        Status-Line  =  SIP-version SP Status-Code SP Reason-Phrase CRLF



<span class="grey">Handley, et al.             Standards Track                    [Page 36]</span></pre>
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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-5.1.1" href="#section-5.1.1">5.1.1</a> Status Codes and Reason Phrases</span>

   The Status-Code is a 3-digit integer result code that indicates the
   outcome of the attempt to understand and satisfy the request. The
   Reason-Phrase is intended to give a short textual description of the
   Status-Code. The Status-Code is intended for use by automata, whereas
   the Reason-Phrase is intended for the human user. The client is not
   required to examine or display the Reason-Phrase.



        Status-Code     =  Informational                     ;Fig. 5
                       |   Success                           ;Fig. 5
                       |   Redirection                       ;Fig. 6
                       |   Client-Error                      ;Fig. 7
                       |   Server-Error                      ;Fig. 8
                       |   Global-Failure                    ;Fig. 9
                       |   extension-code
        extension-code  =  3DIGIT
        Reason-Phrase   =  *&lt;TEXT-UTF8,  excluding CR, LF&gt;


   We provide an overview of the Status-Code below, and provide full
   definitions in <a href="#section-7">Section 7</a>. The first digit of the Status-Code defines
   the class of response. The last two digits do not have any
   categorization role. SIP/2.0 allows 6 values for the first digit:

   1xx: Informational -- request received, continuing to process the
        request;

   2xx: Success -- the action was successfully received, understood, and
        accepted;

   3xx: Redirection -- further action needs to be taken in order to
        complete the request;

   4xx: Client Error -- the request contains bad syntax or cannot be
        fulfilled at this server;

   5xx: Server Error -- the server failed to fulfill an apparently valid
        request;

   6xx: Global Failure -- the request cannot be fulfilled at any server.

   Figures 5 through 9 present the individual values of the numeric
   response codes, and an example set of corresponding reason phrases
   for SIP/2.0. These reason phrases are only recommended; they may be
   replaced by local equivalents without affecting the protocol. Note



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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
   codes in the range starting at x80 to avoid conflicts with newly
   defined HTTP response codes, and adds a new class, 6xx, of response
   codes.

   SIP response codes are extensible. SIP applications are not required
   to understand the meaning of all registered response codes, though
   such understanding is obviously desirable. However, applications MUST
   understand the class of any response code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 response code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if a client
   receives an unrecognized response code of 431, it can safely assume
   that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code. In
   such cases, user agents SHOULD present to the user the message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual
   status.



        Informational  =  "100"  ;  Trying
                      |   "180"  ;  Ringing
                      |   "181"  ;  Call Is Being Forwarded
                      |   "182"  ;  Queued
        Success        =  "200"  ;  OK


   Figure 5: Informational and success status codes





        Redirection  =  "300"  ;  Multiple Choices
                    |   "301"  ;  Moved Permanently
                    |   "302"  ;  Moved Temporarily
                    |   "303"  ;  See Other
                    |   "305"  ;  Use Proxy
                    |   "380"  ;  Alternative Service


   Figure 6: Redirection status codes







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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




        Client-Error  =  "400"  ;  Bad Request
                     |   "401"  ;  Unauthorized
                     |   "402"  ;  Payment Required
                     |   "403"  ;  Forbidden
                     |   "404"  ;  Not Found
                     |   "405"  ;  Method Not Allowed
                     |   "406"  ;  Not Acceptable
                     |   "407"  ;  Proxy Authentication Required
                     |   "408"  ;  Request Timeout
                     |   "409"  ;  Conflict
                     |   "410"  ;  Gone
                     |   "411"  ;  Length Required
                     |   "413"  ;  Request Entity Too Large
                     |   "414"  ;  Request-URI Too Large
                     |   "415"  ;  Unsupported Media Type
                     |   "420"  ;  Bad Extension
                     |   "480"  ;  Temporarily not available
                     |   "481"  ;  Call Leg/Transaction Does Not Exist
                     |   "482"  ;  Loop Detected
                     |   "483"  ;  Too Many Hops
                     |   "484"  ;  Address Incomplete
                     |   "485"  ;  Ambiguous
                     |   "486"  ;  Busy Here


   Figure 7: Client error status codes


        Server-Error  =  "500"  ;  Internal Server Error
                     |   "501"  ;  Not Implemented
                     |   "502"  ;  Bad Gateway
                     |   "503"  ;  Service Unavailable
                     |   "504"  ;  Gateway Time-out
                     |   "505"  ;  SIP Version not supported


   Figure 8: Server error status codes


<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a> Header Field Definitions</span>

   SIP header fields are similar to HTTP header fields in both syntax
   and semantics. In particular, SIP header fields follow the syntax for
   message-header as described in [H4.2]. The rules for extending header
   fields over multiple lines, and use of multiple message-header fields
   with the same field-name, described in [H4.2] also apply to SIP. The



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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




        Global-Failure |  "600"  ;  Busy Everywhere
                       |  "603"  ;  Decline
                       |  "604"  ;  Does not exist anywhere
                       |  "606"  ;  Not Acceptable


   Figure 9: Global failure status codes


   rules in [H4.2] regarding ordering of header fields apply to SIP,
   with the exception of Via fields, see below, whose order matters.
   Additionally, header fields which are hop-by-hop MUST appear before
   any header fields which are end-to-end. Proxies SHOULD NOT reorder
   header fields. Proxies add Via header fields and MAY add other hop-
   by-hop header fields. They can modify certain header fields, such as
   Max-Forwards (<a href="#section-6.23">Section 6.23</a>) and "fix up" the Via header fields with
   "received" parameters as described in <a href="#section-6.40.1">Section 6.40.1</a>. Proxies MUST
   NOT alter any fields that are authenticated (see <a href="#section-13.2">Section 13.2</a>).

   The header fields required, optional and not applicable for each
   method are listed in Table 4 and Table 5. The table uses "o" to
   indicate optional, "m" mandatory and "-" for not applicable. A "*"
   indicates that the header fields are needed only if message body is
   not empty. See sections <a href="#section-6.15">6.15</a>, <a href="#section-6.16">6.16</a> and <a href="#section-8">8</a> for details.

   The "where" column describes the request and response types with
   which the header field can be used. "R" refers to header fields that
   can be used in requests (that is, request and general header fields).
   "r" designates a response or general-header field as applicable to
   all responses, while a list of numeric values indicates the status
   codes with which the header field can be used. "g" and "e" designate
   general (<a href="#section-6.1">Section 6.1</a>) and entity header (<a href="#section-6.2">Section 6.2</a>) fields,
   respectively. If a header field is marked "c", it is copied from the
   request to the response.

   The "enc." column describes whether this message header field MAY be
   encrypted end-to-end. A "n" designates fields that MUST NOT be
   encrypted, while "c" designates fields that SHOULD be encrypted if
   encryption is used.

   The "e-e" column has a value of "e" for end-to-end and a value of "h"
   for hop-by-hop header fields.







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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-41" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>



                          where  enc.  e-e ACK BYE CAN INV OPT REG
        __________________________________________________________
        Accept              R           e   -   -   -   o   o   o
        Accept             415          e   -   -   -   o   o   o
        Accept-Encoding     R           e   -   -   -   o   o   o
        Accept-Encoding    415          e   -   -   -   o   o   o
        Accept-Language     R           e   -   o   o   o   o   o
        Accept-Language    415          e   -   o   o   o   o   o
        Allow              200          e   -   -   -   -   m   -
        Allow              405          e   o   o   o   o   o   o
        Authorization       R           e   o   o   o   o   o   o
        Call-ID            gc     n     e   m   m   m   m   m   m
        Contact             R           e   o   -   -   o   o   o
        Contact            1xx          e   -   -   -   o   o   -
        Contact            2xx          e   -   -   -   o   o   o
        Contact            3xx          e   -   o   -   o   o   o
        Contact            485          e   -   o   -   o   o   o
        Content-Encoding    e           e   o   -   -   o   o   o
        Content-Length      e           e   o   -   -   o   o   o
        Content-Type        e           e   *   -   -   *   *   *
        CSeq               gc     n     e   m   m   m   m   m   m
        Date                g           e   o   o   o   o   o   o
        Encryption          g     n     e   o   o   o   o   o   o
        Expires             g           e   -   -   -   o   -   o
        From               gc     n     e   m   m   m   m   m   m
        Hide                R     n     h   o   o   o   o   o   o
        Max-Forwards        R     n     e   o   o   o   o   o   o
        Organization        g     c     h   -   -   -   o   o   o


   Table 4: Summary of header fields, A--O

   Other header fields can be added as required; a server MUST ignore
   header fields not defined in this specification that it does not
   understand. A proxy MUST NOT remove or modify header fields not
   defined in this specification that it does not understand. A compact
   form of these header fields is also defined in <a href="#section-9">Section 9</a> for use over
   UDP when the request has to fit into a single packet and size is an
   issue.

   Table 6 in <a href="#appendix-A">Appendix A</a> lists those header fields that different client
   and server types MUST be able to parse.

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a> General Header Fields</span>

   General header fields apply to both request and response messages.
   The "general-header" field names can be extended reliably only in
   combination with a change in the protocol version. However, new or


<span class="grey">Handley, et al.             Standards Track                    [Page 41]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-42" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>



                            where     enc.  e-e ACK BYE CAN INV OPT REG
    ___________________________________________________________________
    Proxy-Authenticate       407       n     h   o   o   o   o   o   o
    Proxy-Authorization       R        n     h   o   o   o   o   o   o
    Proxy-Require             R        n     h   o   o   o   o   o   o
    Priority                  R        c     e   -   -   -   o   -   -
    Require                   R              e   o   o   o   o   o   o
    Retry-After               R        c     e   -   -   -   -   -   o
    Retry-After          404,480,486   c     e   o   o   o   o   o   o
                             503       c     e   o   o   o   o   o   o
                           600,603     c     e   o   o   o   o   o   o
    Response-Key              R        c     e   -   o   o   o   o   o
    Record-Route              R              h   o   o   o   o   o   o
    Record-Route             2xx             h   o   o   o   o   o   o
    Route                     R              h   o   o   o   o   o   o
    Server                    r        c     e   o   o   o   o   o   o
    Subject                   R        c     e   -   -   -   o   -   -
    Timestamp                 g              e   o   o   o   o   o   o
    To                      gc(1)      n     e   m   m   m   m   m   m
    Unsupported              420             e   o   o   o   o   o   o
    User-Agent                g        c     e   o   o   o   o   o   o
    Via                     gc(2)      n     e   m   m   m   m   m   m
    Warning                   r              e   o   o   o   o   o   o
    WWW-Authenticate         401       c     e   o   o   o   o   o   o


   Table 5: Summary of header fields, P--Z; (1):  copied  with  possible
   addition of tag; (2): UAS removes first Via header field

   experimental header fields MAY be given the semantics of general
   header fields if all parties in the communication recognize them to
   be "general-header" fields. Unrecognized header fields are treated as
   "entity-header" fields.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a> Entity Header Fields</span>

   The "entity-header" fields define meta-information about the
   message-body or, if no body is present, about the resource identified
   by the request. The term "entity header" is an HTTP 1.1 term where
   the response body can contain a transformed version of the message
   body.  The original message body is referred to as the "entity". We
   retain the same terminology for header fields but usually refer to
   the "message body" rather then the entity as the two are the same in
   SIP.






<span class="grey">Handley, et al.             Standards Track                    [Page 42]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-43" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a> Request Header Fields</span>

   The "request-header" fields allow the client to pass additional
   information about the request, and about the client itself, to the
   server. These fields act as request modifiers, with semantics
   equivalent to the parameters of a programming language method
   invocation.

   The "request-header" field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of "request-
   header" fields if all parties in the communication recognize them to
   be request-header fields. Unrecognized header fields are treated as
   "entity-header" fields.

<span class="h3"><a class="selflink" id="section-6.4" href="#section-6.4">6.4</a> Response Header Fields</span>

   The "response-header" fields allow the server to pass additional
   information about the response which cannot be placed in the Status-
   Line. These header fields give information about the server and about
   further access to the resource identified by the Request-URI.

   Response-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of "response-
   header" fields if all parties in the communication recognize them to
   be "response-header" fields. Unrecognized header fields are treated
   as "entity-header" fields.

<span class="h3"><a class="selflink" id="section-6.5" href="#section-6.5">6.5</a> End-to-end and Hop-by-hop Headers</span>

   End-to-end headers MUST be transmitted unmodified across all proxies,
   while hop-by-hop headers MAY be modified or added by proxies.

<span class="h3"><a class="selflink" id="section-6.6" href="#section-6.6">6.6</a> Header Field Format</span>

   Header fields ("general-header", "request-header", "response-header",
   and "entity-header") follow the same generic header format as that
   given in <a href="./rfc822#section-3.1">Section&nbsp;3.1 of RFC 822</a> [<a href="#ref-24" title="&quot;Standard for the format of ARPA internet text messages&quot;">24</a>]. Each header field consists of a
   name followed by a colon (":") and the field value. Field names are
   case-insensitive. The field value MAY be preceded by any amount of
   leading white space (LWS), though a single space (SP) is preferred.
   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or horizontal tab (HT). Applications
   MUST follow HTTP "common form" when generating these constructs,
   since there might exist some implementations that fail to accept
   anything beyond the common forms.




<span class="grey">Handley, et al.             Standards Track                    [Page 43]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-44" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        message-header  =  field-name ":" [ field-value ] CRLF
        field-name      =  token
        field-value     =  *( field-content | LWS )
        field-content   =  &lt; the OCTETs  making up the field-value
                            and consisting of either *TEXT-UTF8
                            or combinations of token,
                            separators, and quoted-string&gt;


   The relative order of header fields with different field names is not
   significant. Multiple header fields with the same field-name may be
   present in a message if and only if the entire field-value for that
   header field is defined as a comma-separated list (i.e., #(values)).
   It MUST be possible to combine the multiple header fields into one
   "field-name: field-value" pair, without changing the semantics of the
   message, by appending each subsequent field-value to the first, each
   separated by a comma. The order in which header fields with the same
   field-name are received is therefore significant to the
   interpretation of the combined field value, and thus a proxy MUST NOT
   change the order of these field values when a message is forwarded.

   Field names are not case-sensitive, although their values may be.

<span class="h3"><a class="selflink" id="section-6.7" href="#section-6.7">6.7</a> Accept</span>

   The Accept header follows the syntax defined in [H14.1]. The
   semantics are also identical, with the exception that if no Accept
   header is present, the server SHOULD assume a default value of
   application/sdp.

   This request-header field is used only with the INVITE, OPTIONS and
   REGISTER request methods to indicate what media types are acceptable
   in the response.

   Example:


     Accept: application/sdp;level=1, application/x-private, text/html



<span class="h3"><a class="selflink" id="section-6.8" href="#section-6.8">6.8</a> Accept-Encoding</span>

   The Accept-Encoding request-header field is similar to Accept, but
   restricts the content-codings [H3.4.1] that are acceptable in the
   response. See [H14.3]. The syntax of this header is defined in
   [H14.3]. The semantics in SIP are identical to those defined in
   [H14.3].



<span class="grey">Handley, et al.             Standards Track                    [Page 44]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-45" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.9" href="#section-6.9">6.9</a> Accept-Language</span>

   The Accept-Language header follows the syntax defined in [H14.4]. The
   rules for ordering the languages based on the q parameter apply to
   SIP as well. When used in SIP, the Accept-Language request-header
   field can be used to allow the client to indicate to the server in
   which language it would prefer to receive reason phrases, session
   descriptions or status responses carried as message bodies. A proxy
   MAY use this field to help select the destination for the call, for
   example, a human operator conversant in a language spoken by the
   caller.

   Example:


     Accept-Language: da, en-gb;q=0.8, en;q=0.7


<span class="h3"><a class="selflink" id="section-6.10" href="#section-6.10">6.10</a> Allow</span>

   The Allow entity-header field lists the set of methods supported by
   the resource identified by the Request-URI. The purpose of this field
   is strictly to inform the recipient of valid methods associated with
   the resource. An Allow header field MUST be present in a 405 (Method
   Not Allowed) response and SHOULD be present in an OPTIONS response.



        Allow  =  "Allow" ":" 1#Method


<span class="h3"><a class="selflink" id="section-6.11" href="#section-6.11">6.11</a> Authorization</span>

   A user agent that wishes to authenticate itself with a server --
   usually, but not necessarily, after receiving a 401 response -- MAY
   do so by including an Authorization request-header field with the
   request. The Authorization field value consists of credentials
   containing the authentication information of the user agent for the
   realm of the resource being requested.

   <a href="#section-13.2">Section 13.2</a> overviews the use of the Authorization header, and
   <a href="#section-15">section 15</a> describes the syntax and semantics when used with PGP
   based authentication.








<span class="grey">Handley, et al.             Standards Track                    [Page 45]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-46" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.12" href="#section-6.12">6.12</a> Call-ID</span>

   The Call-ID general-header field uniquely identifies a particular
   invitation or all registrations of a particular client. Note that a
   single multimedia conference can give rise to several calls with
   different Call-IDs, e.g., if a user invites a single individual
   several times to the same (long-running) conference.

   For an INVITE request, a callee user agent server SHOULD NOT alert
   the user if the user has responded previously to the Call-ID in the
   INVITE request. If the user is already a member of the conference and
   the conference parameters contained in the session description have
   not changed, a callee user agent server MAY silently accept the call,
   regardless of the Call-ID. An invitation for an existing Call-ID or
   session can change the parameters of the conference. A client
   application MAY decide to simply indicate to the user that the
   conference parameters have been changed and accept the invitation
   automatically or it MAY require user confirmation.

   A user may be invited to the same conference or call using several
   different Call-IDs. If desired, the client MAY use identifiers within
   the session description to detect this duplication. For example, SDP
   contains a session id and version number in the origin (o) field.

   The REGISTER and OPTIONS methods use the Call-ID value to
   unambiguously match requests and responses. All REGISTER requests
   issued by a single client SHOULD use the same Call-ID, at least
   within the same boot cycle.


        Since the Call-ID is generated by and for SIP, there is no
        reason to deal with the complexity of URL-encoding and
        case-ignoring string comparison.



        Call-ID   =  ( "Call-ID" | "i" ) ":" local-id "@" host
        local-id  =  1*uric


   "host" SHOULD be either a fully qualified domain name or a globally
   routable IP address. If this is the case, the "local-id" SHOULD be an
   identifier consisting of URI characters that is unique within "host".
   Use of cryptographically random identifiers [<a href="#ref-27" title="&quot;Randomness recommendations for security&quot;">27</a>] is RECOMMENDED.  If,
   however, host is not an FQDN or globally routable IP address (such as
   a net 10 address), the local-id MUST be globally unique, as opposed





<span class="grey">Handley, et al.             Standards Track                    [Page 46]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-47" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   to unique within host. These rules guarantee overall global
   uniqueness of the Call-ID. The value for Call-ID MUST NOT be reused
   for a different call.  Call-IDs are case-sensitive.


        Using cryptographically random identifiers provides some
        protection against session hijacking. Call-ID, To and From
        are needed to identify a call leg.  The distinction between
        call and call leg matters in calls with third-party
        control.

   For systems which have tight bandwidth constraints, many of the
   mandatory SIP headers have a compact form, as discussed in <a href="#section-9">Section 9</a>.
   These are alternate names for the headers which occupy less space in
   the message. In the case of Call-ID, the compact form is i.

   For example, both of the following are valid:

     Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com


   or

     i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com



<span class="h3"><a class="selflink" id="section-6.13" href="#section-6.13">6.13</a> Contact</span>

   The Contact general-header field can appear in INVITE, ACK, and
   REGISTER requests, and in 1xx, 2xx, 3xx, and 485 responses. In
   general, it provides a URL where the user can be reached for further
   communications.

   INVITE and ACK requests: INVITE and ACK requests MAY contain Contact
        headers indicating from which location the request is
        originating.


        This allows the callee to send future requests, such as
        BYE, directly to the caller instead of through a series of
        proxies.  The Via header is not sufficient since the
        desired address may be that of a proxy.

   INVITE 2xx responses: A user agent server sending a definitive,
        positive response (2xx) MAY insert a Contact response header
        field indicating the SIP address under which it is reachable
        most directly for future SIP requests, such as ACK, within the



<span class="grey">Handley, et al.             Standards Track                    [Page 47]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-48" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        same Call-ID. The Contact header field contains the address of
        the server itself or that of a proxy, e.g., if the host is
        behind a firewall. The value of this Contact header is copied
        into the Request-URI of subsequent requests for this call if the
        response did not also contain a Record-Route header. If the
        response also contains a Record-Route header field, the address
        in the Contact header field is added as the last item in the
        Route header field. See <a href="#section-6.29">Section 6.29</a> for details.


        The Contact value SHOULD NOT be cached across calls, as it
        may not represent the most desirable location for a
        particular destination address.

   INVITE 1xx responses: A UAS sending a provisional response (1xx) MAY
        insert a Contact response header. It has the same semantics in a
        1xx response as a 2xx INVITE response. Note that CANCEL requests
        MUST NOT be sent to that address, but rather follow the same
        path as the original request.

   REGISTER requests: REGISTER requests MAY contain a Contact header
        field indicating at which locations the user is reachable. The
        REGISTER request defines a wildcard Contact field, "*", which
        MUST only be used with Expires: 0 to remove all registrations
        for a particular user. An optional "expires" parameter indicates
        the desired expiration time of the registration. If a Contact
        entry does not have an "expires" parameter, the Expires header
        field is used as the default value. If neither of these
        mechanisms is used, SIP URIs are assumed to expire after one
        hour. Other URI schemes have no expiration times.

   REGISTER 2xx responses: A REGISTER response MAY return all locations
        at which the user is currently reachable.  An optional "expires"
        parameter indicates the expiration time of the registration. If
        a Contact entry does not have an "expires" parameter, the value
        of the Expires header field indicates the expiration time. If
        neither mechanism is used, the expiration time specified in the
        request, explicitly or by default, is used.

   3xx and 485 responses: The Contact response-header field can be used
        with a 3xx or 485 (Ambiguous) response codes to indicate one or
        more alternate addresses to try. It can appear in responses to
        BYE, INVITE and OPTIONS methods. The Contact header field
        contains URIs giving the new locations or user names to try, or
        may simply specify additional transport parameters. A 300
        (Multiple Choices), 301 (Moved Permanently), 302 (Moved
        Temporarily) or 485 (Ambiguous) response SHOULD contain a
        Contact field containing URIs of new addresses to be tried. A



<span class="grey">Handley, et al.             Standards Track                    [Page 48]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-50" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        301 or 302 response may also give the same location and username
        that was being tried but specify additional transport parameters
        such as a different server or multicast address to try or a
        change of SIP transport from UDP to TCP or vice versa. The
        client copies the "user", "password", "host", "port" and "user-
        param" elements of the Contact URI into the Request-URI of the
        redirected request and directs the request to the address
        specified by the "maddr" and "port" parameters, using the
        transport protocol given in the "transport" parameter. If
        "maddr" is a multicast address, the value of "ttl" is used as
        the time-to-live value.

   Note that the Contact header field MAY also refer to a different
   entity than the one originally called. For example, a SIP call
   connected to GSTN gateway may need to deliver a special information
   announcement such as "The number you have dialed has been changed."

   A Contact response header field can contain any suitable URI
   indicating where the called party can be reached, not limited to SIP
   URLs. For example, it could contain URL's for phones, fax, or irc (if
   they were defined) or a mailto: (<a href="./rfc2368">RFC 2368</a>, [<a href="#ref-28" title="&quot;The mailto URL scheme&quot;">28</a>]) URL.

   The following parameters are defined. Additional parameters may be
   defined in other specifications.

   q: The "qvalue" indicates the relative preference among the locations
        given. "qvalue" values are decimal numbers from 0 to 1, with
        higher values indicating higher preference.

   action: The "action" parameter is used only when registering with the
        REGISTER request. It indicates whether the client wishes that
        the server proxy or redirect future requests intended for the
        client. If this parameter is not specified the action taken
        depends on server configuration. In its response, the registrar
        SHOULD indicate the mode used. This parameter is ignored for
        other requests.

   expires: The "expires" parameter indicates how long the URI is valid.
        The parameter is either a number indicating seconds or a quoted
        string containing a SIP-date. If this parameter is not provided,
        the value of the Expires header field determines how long the
        URI is valid. Implementations MAY treat values larger than
        2**32-1 (4294967295 seconds or 136 years) as equivalent to
        2**32-1.



   Contact = ( "Contact" | "m" ) ":"
             ("*" | (1# (( name-addr | addr-spec )
             [ *( ";" contact-params ) ] [ comment ] )))

   name-addr      = [ display-name ] "&lt;" addr-spec "&gt;"
   addr-spec      = SIP-URL | URI
   display-name   = *token | quoted-string

   contact-params = "q"       "=" qvalue
                  | "action"  "=" "proxy" | "redirect"
                  | "expires" "=" delta-seconds | &lt;"&gt; SIP-date &lt;"&gt;
                  | extension-attribute

   extension-attribute = extension-name [ "=" extension-value ]

        only allows one address, unquoted. Since URIs can contain
        commas and semicolons as reserved characters, they can be
        mistaken for header or parameter delimiters, respectively.
        The current syntax corresponds to that for the To and From
        header, which also allows the use of display names.

   Example:


     Contact: "Mr. Watson" &lt;sip:watson@worcester.bell-telephone.com&gt;
        ;q=0.7; expires=3600,
        "Mr. Watson" &lt;mailto:watson@bell-telephone.com&gt; ;q=0.1



<span class="h3"><a class="selflink" id="section-6.14" href="#section-6.14">6.14</a> Content-Encoding</span>



        Content-Encoding  =  ( "Content-Encoding" | "e" ) ":"
                             1#content-coding


   The Content-Encoding entity-header field is used as a modifier to the
   "media-type". When present, its value indicates what additional
   content codings have been applied to the entity-body, and thus what
   decoding mechanisms MUST be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a body to be compressed without losing the
   identity of its underlying media type.

   If multiple encodings have been applied to an entity, the content
   codings MUST be listed in the order in which they were applied.

   All content-coding values are case-insensitive. The Internet Assigned
   Numbers Authority (IANA) acts as a registry for content-coding value
   tokens. See [3.5] for a definition of the syntax for content-coding.

   Clients MAY apply content encodings to the body in requests. If the
   server is not capable of decoding the body, or does not recognize any
   of the content-coding values, it MUST send a 415 "Unsupported Media
   Type" response, listing acceptable encodings in the Accept-Encoding



<span class="grey">Handley, et al.             Standards Track                    [Page 50]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-51" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   header. A server MAY apply content encodings to the bodies in
   responses. The server MUST only use encodings listed in the Accept-
   Encoding header in the request.

<span class="h3"><a class="selflink" id="section-6.15" href="#section-6.15">6.15</a> Content-Length</span>

   The Content-Length entity-header field indicates the size of the
   message-body, in decimal number of octets, sent to the recipient.



        Content-Length  =  ( "Content-Length" | "l" ) ":" 1*DIGIT


   An example is

     Content-Length: 3495



   Applications SHOULD use this field to indicate the size of the
   message-body to be transferred, regardless of the media type of the
   entity. Any Content-Length greater than or equal to zero is a valid
   value. If no body is present in a message, then the Content-Length
   header field MUST be set to zero. If a server receives a UDP request
   without Content-Length, it MUST assume that the request encompasses
   the remainder of the packet.  If a server receives a UDP request with
   a Content-Length, but the value is larger than the size of the body
   sent in the request, the client SHOULD generate a 400 class response.
   If there is additional data in the UDP packet after the last byte of
   the body has been read, the server MUST treat the remaining data as a
   separate message. This allows several messages to be placed in a
   single UDP packet.

   If a response does not contain a Content-Length, the client assumes
   that it encompasses the remainder of the UDP packet or the data until
   the TCP connection is closed, as applicable.  <a href="#section-8">Section 8</a> describes how
   to determine the length of the message body.

<span class="h3"><a class="selflink" id="section-6.16" href="#section-6.16">6.16</a> Content-Type</span>

   The Content-Type entity-header field indicates the media type of the
   message-body sent to the recipient. The "media-type" element is
   defined in [H3.7].



        Content-Type  =  ( "Content-Type" | "c" ) ":" media-type



<span class="grey">Handley, et al.             Standards Track                    [Page 51]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-52" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Examples of this header field are

     Content-Type: application/sdp
     Content-Type: text/html; charset=ISO-8859-4



<span class="h3"><a class="selflink" id="section-6.17" href="#section-6.17">6.17</a> CSeq</span>

   Clients MUST add the CSeq (command sequence) general-header field to
   every request. A CSeq header field in a request contains the request
   method and a single decimal sequence number chosen by the requesting
   client, unique within a single value of Call-ID. The sequence number
   MUST be expressible as a 32-bit unsigned integer. The initial value
   of the sequence number is arbitrary, but MUST be less than 2**31.
   Consecutive requests that differ in request method, headers or body,
   but have the same Call-ID MUST contain strictly monotonically
   increasing and contiguous sequence numbers; sequence numbers do not
   wrap around.  Retransmissions of the same request carry the same
   sequence number, but an INVITE with a different message body or
   different header fields (a "re-invitation") acquires a new, higher
   sequence number. A server MUST echo the CSeq value from the request
   in its response.  If the Method value is missing in the received CSeq
   header field, the server fills it in appropriately.

   The ACK and CANCEL requests MUST contain the same CSeq value as the
   INVITE request that it refers to, while a BYE request cancelling an
   invitation MUST have a higher sequence number. A BYE request with a
   CSeq that is not higher should cause a 400 response to be generated.

   A user agent server MUST remember the highest sequence number for any
   INVITE request with the same Call-ID value. The server MUST respond
   to, and then discard, any INVITE request with a lower sequence
   number.

   All requests spawned in a parallel search have the same CSeq value as
   the request triggering the parallel search.



        CSeq  =  "CSeq" ":" 1*DIGIT Method



        Strictly speaking, CSeq header fields are needed for any
        SIP request that can be cancelled by a BYE or CANCEL
        request or where a client can issue several requests for
        the same Call-ID in close succession. Without a sequence



<span class="grey">Handley, et al.             Standards Track                    [Page 52]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-53" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        number, the response to an INVITE could be mistaken for the
        response to the cancellation (BYE or CANCEL). Also, if the
        network duplicates packets or if an ACK is delayed until
        the server has sent an additional response, the client
        could interpret an old response as the response to a re-
        invitation issued shortly thereafter. Using CSeq also makes
        it easy for the server to distinguish different versions of
        an invitation, without comparing the message body.

   The Method value allows the client to distinguish the response to an
   INVITE request from that of a CANCEL response. CANCEL requests can be
   generated by proxies; if they were to increase the sequence number,
   it might conflict with a later request issued by the user agent for
   the same call.

   With a length of 32 bits, a server could generate, within a single
   call, one request a second for about 136 years before needing to wrap
   around.  The initial value of the sequence number is chosen so that
   subsequent requests within the same call will not wrap around. A
   non-zero initial value allows to use a time-based initial sequence
   number, if the client desires. A client could, for example, choose
   the 31 most significant bits of a 32-bit second clock as an initial
   sequence number.

   Forked requests MUST have the same CSeq as there would be ambiguity
   otherwise between these forked requests and later BYE issued by the
   client user agent.

   Example:


     CSeq: 4711 INVITE



<span class="h3"><a class="selflink" id="section-6.18" href="#section-6.18">6.18</a> Date</span>

   Date is a general-header field. Its syntax is:



        SIP-date  =  <a href="./rfc1123">rfc1123</a>-date


   See [H14.19] for a definition of <a href="./rfc1123">rfc1123</a>-date. Note that unlike
   HTTP/1.1, SIP only supports the most recent <a href="./rfc1123">RFC1123</a> [<a href="#ref-29" title="&quot;Requirements for internet hosts - application and support&quot;">29</a>] formatting
   for dates.




<span class="grey">Handley, et al.             Standards Track                    [Page 53]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-54" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The Date header field reflects the time when the request or response
   is first sent. Thus, retransmissions have the same Date header field
   value as the original.


        The Date header field can be used by simple end systems
        without a battery-backed clock to acquire a notion of
        current time.

<span class="h3"><a class="selflink" id="section-6.19" href="#section-6.19">6.19</a> Encryption</span>

   The Encryption general-header field specifies that the content has
   been encrypted. <a href="#section-13">Section 13</a> describes the overall SIP security
   architecture and algorithms. This header field is intended for end-
   to-end encryption of requests and responses. Requests are encrypted
   based on the public key belonging to the entity named in the To
   header field. Responses are encrypted based on the public key
   conveyed in the Response-Key header field. Note that the public keys
   themselves may not be used for the encryption. This depends on the
   particular algorithms used.

   For any encrypted message, at least the message body and possibly
   other message header fields are encrypted. An application receiving a
   request or response containing an Encryption header field decrypts
   the body and then concatenates the plaintext to the request line and
   headers of the original message. Message headers in the decrypted
   part completely replace those with the same field name in the
   plaintext part.  (Note: If only the body of the message is to be
   encrypted, the body has to be prefixed with CRLF to allow proper
   concatenation.) Note that the request method and Request-URI cannot
   be encrypted.


        Encryption only provides privacy; the recipient has no
        guarantee that the request or response came from the party
        listed in the From message header, only that the sender
        used the recipient's public key. However, proxies will not
        be able to modify the request or response.



        Encryption         =  "Encryption" ":" encryption-scheme 1*SP
                              #encryption-params
        encryption-scheme  =  token
        encryption-params  =  token "=" ( token | quoted-string )

        The token indicates the form of encryption used; it is
        described in <a href="#section-13">section 13</a>.



<span class="grey">Handley, et al.             Standards Track                    [Page 54]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-55" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The example in Figure 10 shows a message encrypted with ASCII-armored
   PGP that was generated by applying "pgp -ea" to the payload to be
   encrypted.


   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   From: &lt;sip:a.g.bell@bell-telephone.com&gt;
   To: T. A. Watson &lt;sip:watson@bell-telephone.com&gt;
   Call-ID: 187602141351@worcester.bell-telephone.com
   Content-Length: 885
   Encryption: PGP version=2.6.2,encoding=ascii

   hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red
   h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs
   ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR
   RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA
   zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu
   X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6
   IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/
   GgKADYe4M3JSwOtqwU8zUJF3FIfk7vsxmSqtUQrRQaiIhqNyG7KxJt4YjWnEjF5E
   WUIPhvyGFMJaeQXIyGRYZAYvKKklyAJcm29zLACxU5alX4M25lHQd9FR9Zmq6Jed
   wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO
   z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+
   6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2
   z8X9N4MhMyXEVuC9rt8/AUhmVQ==
   =bOW+



   Figure 10: PGP Encryption Example



   Since proxies can base their forwarding decision on any combination
   of SIP header fields, there is no guarantee that an encrypted request
   "hiding" header fields will reach the same destination as an
   otherwise identical un-encrypted request.

<span class="h3"><a class="selflink" id="section-6.20" href="#section-6.20">6.20</a> Expires</span>

   The Expires entity-header field gives the date and time after which
   the message content expires.

   This header field is currently defined only for the REGISTER and
   INVITE methods. For REGISTER, it is a request and response-header
   field. In a REGISTER request, the client indicates how long it wishes
   the registration to be valid. In the response, the server indicates



<span class="grey">Handley, et al.             Standards Track                    [Page 55]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-56" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   the earliest expiration time of all registrations. The server MAY
   choose a shorter time interval than that requested by the client, but
   SHOULD NOT choose a longer one.

   For INVITE requests, it is a request and response-header field. In a
   request, the caller can limit the validity of an invitation, for
   example, if a client wants to limit the time duration of a search or
   a conference invitation. A user interface MAY take this as a hint to
   leave the invitation window on the screen even if the user is not
   currently at the workstation. This also limits the duration of a
   search. If the request expires before the search completes, the proxy
   returns a 408 (Request Timeout) status. In a 302 (Moved Temporarily)
   response, a server can advise the client of the maximal duration of
   the redirection.

   The value of this field can be either a SIP-date or an integer number
   of seconds (in decimal), measured from the receipt of the request.
   The latter approach is preferable for short durations, as it does not
   depend on clients and servers sharing a synchronized clock.
   Implementations MAY treat values larger than 2**32-1 (4294967295 or
   136 years) as equivalent to 2**32-1.



        Expires  =  "Expires" ":" ( SIP-date | delta-seconds )


   Two examples of its use are

     Expires: Thu, 01 Dec 1994 16:00:00 GMT
     Expires: 5



<span class="h3"><a class="selflink" id="section-6.21" href="#section-6.21">6.21</a> From</span>

   Requests and responses MUST contain a From general-header field,
   indicating the initiator of the request. The From field MAY contain
   the "tag" parameter. The server copies the From header field from the
   request to the response. The optional "display-name" is meant to be
   rendered by a human-user interface. A system SHOULD use the display
   name "Anonymous" if the identity of the client is to remain hidden.

   The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
   "ttl-param", or "headers" elements. A server that receives a SIP-URL
   with these elements removes them before further processing.





<span class="grey">Handley, et al.             Standards Track                    [Page 56]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-57" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.



        From         =  ( "From" | "f" ) ":" ( name-addr | addr-spec )
                        *( ";" addr-params )
        addr-params  =  tag-param
        tag-param    =  "tag=" UUID
        UUID         =  1*( hex | "-" )


   Examples:


     From: "A. G. Bell" &lt;sip:agb@bell-telephone.com&gt;
     From: sip:+12125551212@server.phone2net.com
     From: Anonymous &lt;sip:c8oqz84zk7z@privacy.org&gt;



   The "tag" MAY appear in the From field of a request. It MUST be
   present when it is possible that two instances of a user sharing a
   SIP address can make call invitations with the same Call-ID.

   The "tag" value MUST be globally unique and cryptographically random
   with at least 32 bits of randomness. A single user maintains the same
   tag throughout the call identified by the Call-ID.


        Call-ID, To and From are needed to identify a call leg.
        The distinction between call and call leg matters in calls
        with multiple responses to a forked request. The format is
        similar to the equivalent <a href="./rfc822">RFC 822</a> [<a href="#ref-24" title="&quot;Standard for the format of ARPA internet text messages&quot;">24</a>] header, but with a
        URI instead of just an email address.

<span class="h3"><a class="selflink" id="section-6.22" href="#section-6.22">6.22</a> Hide</span>

   A client uses the Hide request header field to indicate that it wants
   the path comprised of the Via header fields (<a href="#section-6.40">Section 6.40</a>) to be
   hidden from subsequent proxies and user agents. It can take two
   forms: Hide: route and Hide:  hop. Hide header fields are typically
   added by the client user agent, but MAY be added by any proxy along
   the path.






<span class="grey">Handley, et al.             Standards Track                    [Page 57]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-58" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   If a request contains the "Hide: route" header field, all following
   proxies SHOULD hide their previous hop. If a request contains the
   "Hide: hop" header field, only the next proxy SHOULD hide the
   previous hop and then remove the Hide option unless it also wants to
   remain anonymous.

   A server hides the previous hop by encrypting the "host" and "port"
   parts of the top-most Via header field with an algorithm of its
   choice. Servers SHOULD add additional "salt" to the "host" and "port"
   information prior to encryption to prevent malicious downstream
   proxies from guessing earlier parts of the path based on seeing
   identical encrypted Via headers. Hidden Via fields are marked with
   the "hidden" Via option, as described in <a href="#section-6.40">Section 6.40</a>.

   A server that is capable of hiding Via headers MUST attempt to
   decrypt all Via headers marked as "hidden" to perform loop detection.
   Servers that are not capable of hiding can ignore hidden Via fields
   in their loop detection algorithm.


        If hidden headers were not marked, a proxy would have to
        decrypt all headers to detect loops, just in case one was
        encrypted, as the Hide: Hop option may have been removed
        along the way.

   A host MUST NOT add such a "Hide: hop" header field unless it can
   guarantee it will only send a request for this destination to the
   same next hop. The reason for this is that it is possible that the
   request will loop back through this same hop from a downstream proxy.
   The loop will be detected by the next hop if the choice of next hop
   is fixed, but could loop an arbitrary number of times otherwise.

   A client requesting "Hide: route" can only rely on keeping the
   request path private if it sends the request to a trusted proxy.
   Hiding the route of a SIP request is of limited value if the request
   results in data packets being exchanged directly between the calling
   and called user agent.

   The use of Hide header fields is discouraged unless path privacy is
   truly needed; Hide fields impose extra processing costs and
   restrictions for proxies and can cause requests to generate 482 (Loop
   Detected) responses that could otherwise be avoided.

   The encryption of Via header fields is described in more detail in
   <a href="#section-13">Section 13</a>.

   The Hide header field has the following syntax:




<span class="grey">Handley, et al.             Standards Track                    [Page 58]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-59" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Hide  =  "Hide" ":" ( "route" | "hop" )


<span class="h3"><a class="selflink" id="section-6.23" href="#section-6.23">6.23</a> Max-Forwards</span>

   The Max-Forwards request-header field may be used with any SIP method
   to limit the number of proxies or gateways that can forward the
   request to the next downstream server. This can also be useful when
   the client is attempting to trace a request chain which appears to be
   failing or looping in mid-chain.



        Max-Forwards  =  "Max-Forwards" ":" 1*DIGIT


   The Max-Forwards value is a decimal integer indicating the remaining
   number of times this request message is allowed to be forwarded.

   Each proxy or gateway recipient of a request containing a Max-
   Forwards header field MUST check and update its value prior to
   forwarding the request. If the received value is zero (0), the
   recipient MUST NOT forward the request. Instead, for the OPTIONS and
   REGISTER methods, it MUST respond as the final recipient. For all
   other methods, the server returns 483 (Too many hops).

   If the received Max-Forwards value is greater than zero, then the
   forwarded message MUST contain an updated Max-Forwards field with a
   value decremented by one (1).

   Example:

     Max-Forwards: 6



<span class="h3"><a class="selflink" id="section-6.24" href="#section-6.24">6.24</a> Organization</span>

   The Organization general-header field conveys the name of the
   organization to which the entity issuing the request or response
   belongs. It MAY also be inserted by proxies at the boundary of an
   organization.


        The field MAY be used by client software to filter calls.






<span class="grey">Handley, et al.             Standards Track                    [Page 59]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-60" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Organization  =  "Organization" ":" *TEXT-UTF8


<span class="h3"><a class="selflink" id="section-6.25" href="#section-6.25">6.25</a> Priority</span>

   The Priority request-header field indicates the urgency of the
   request as perceived by the client.



        Priority        =  "Priority" ":" priority-value
        priority-value  =  "emergency" | "urgent" | "normal"
                        |  "non-urgent"


   It is RECOMMENDED that the value of "emergency" only be used when
   life, limb or property are in imminent danger.

   Examples:


     Subject: A tornado is heading our way!
     Priority: emergency

     Subject: Weekend plans
     Priority: non-urgent




        These are the values of <a href="./rfc2076">RFC 2076</a> [<a href="#ref-30" title="&quot;Common internet message headers&quot;">30</a>], with the addition of
        "emergency".

<span class="h3"><a class="selflink" id="section-6.26" href="#section-6.26">6.26</a> Proxy-Authenticate</span>

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response. The field value
   consists of a challenge that indicates the authentication scheme and
   parameters applicable to the proxy for this Request-URI.

   Unlike its usage within HTTP, the Proxy-Authenticate header MUST be
   passed upstream in the response to the UAC. In SIP, only UAC's can
   authenticate themselves to proxies.

   The syntax for this header is defined in [H14.33]. See 14 for further
   details on its usage.





<span class="grey">Handley, et al.             Standards Track                    [Page 60]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-61" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   A client SHOULD cache the credentials used for a particular proxy
   server and realm for the next request to that server. Credentials
   are, in general, valid for a specific value of the Request-URI at a
   particular proxy server. If a client contacts a proxy server that has
   required authentication in the past, but the client does not have
   credentials for the particular Request-URI, it MAY attempt to use the
   most-recently used credential. The server responds with 401
   (Unauthorized) if the client guessed wrong.


        This suggested caching behavior is motivated by proxies
        restricting phone calls to authenticated users. It seems
        likely that in most cases, all destinations require the
        same password. Note that end-to-end authentication is
        likely to be destination-specific.

<span class="h3"><a class="selflink" id="section-6.27" href="#section-6.27">6.27</a> Proxy-Authorization</span>

   The Proxy-Authorization request-header field allows the client to
   identify itself (or its user) to a proxy which requires
   authentication. The Proxy-Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the proxy and/or realm of the resource being requested.

   Unlike Authorization, the Proxy-Authorization header field applies
   only to the next outbound proxy that demanded authentication using
   the Proxy- Authenticate field. When multiple proxies are used in a
   chain, the Proxy-Authorization header field is consumed by the first
   outbound proxy that was expecting to receive credentials. A proxy MAY
   relay the credentials from the client request to the next proxy if
   that is the mechanism by which the proxies cooperatively authenticate
   a given request.

   See [H14.34] for a definition of the syntax, and <a href="#section-14">section 14</a> for a
   discussion of its usage.

<span class="h3"><a class="selflink" id="section-6.28" href="#section-6.28">6.28</a> Proxy-Require</span>

   The Proxy-Require header field is used to indicate proxy-sensitive
   features that MUST be supported by the proxy. Any Proxy-Require
   header field features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client if not supported.
   Proxy servers treat this field identically to the Require field.

   See <a href="#section-6.30">Section 6.30</a> for more details on the mechanics of this message
   and a usage example.





<span class="grey">Handley, et al.             Standards Track                    [Page 61]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-62" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.29" href="#section-6.29">6.29</a> Record-Route</span>

   The Record-Route request and response header field is added to a
   request by any proxy that insists on being in the path of subsequent
   requests for the same call leg. It contains a globally reachable
   Request-URI that identifies the proxy server. Each proxy server adds
   its Request-URI to the beginning of the list.

   The server copies the Record-Route header field unchanged into the
   response. (Record-Route is only relevant for 2xx responses.)

   The calling user agent client copies the Record-Route header into a
   Route header field of subsequent requests within the same call leg,
   reversing the order of requests, so that the first entry is closest
   to the user agent client. If the response contained a Contact header
   field, the calling user agent adds its content as the last Route
   header. Unless this would cause a loop, any client MUST send any
   subsequent requests for this call leg to the first Request-URI in the
   Route request header field and remove that entry.

   The calling user agent MUST NOT use the Record-Route header field in
   requests that contain Route header fields.


        Some proxies, such as those controlling firewalls or in an
        automatic call distribution (ACD) system, need to maintain
        call state and thus need to receive any BYE and ACK packets
        for the call.

   The Record-Route header field has the following syntax:


        Record-Route  =  "Record-Route" ":" 1# name-addr


   Proxy servers SHOULD use the "maddr" URL parameter containing their
   address to ensure that subsequent requests are guaranteed to reach
   exactly the same server.

   Example for a request that has traversed the hosts ieee.org and
   bell-telephone.com , in that order:

     Record-Route: &lt;sip:a.g.bell@bell-telephone.com&gt;,
       &lt;sip:a.bell@ieee.org&gt;







<span class="grey">Handley, et al.             Standards Track                    [Page 62]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-63" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.30" href="#section-6.30">6.30</a> Require</span>

   The Require request-header field is used by clients to tell user
   agent servers about options that the client expects the server to
   support in order to properly process the request. If a server does
   not understand the option, it MUST respond by returning status code
   420 (Bad Extension) and list those options it does not understand in
   the Unsupported header.



        Require  =  "Require" ":" 1#option-tag


   Example:

   C-&gt;S:   INVITE sip:watson@bell-telephone.com SIP/2.0
           Require: com.example.billing
           Payment: sheep_skins, conch_shells

   S-&gt;C:   SIP/2.0 420 Bad Extension
           Unsupported: com.example.billing



        This is to make sure that the client-server interaction
        will proceed without delay when all options are understood
        by both sides, and only slow down if options are not
        understood (as in the example above).  For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes ambiguity when the
        client requires features that the server does not
        understand. Some features, such as call handling fields,
        are only of interest to end systems.

   Proxy and redirect servers MUST ignore features that are not
   understood. If a particular extension requires that intermediate
   devices support it, the extension MUST be tagged in the Proxy-Require
   field as well (see <a href="#section-6.28">Section 6.28</a>).

<span class="h3"><a class="selflink" id="section-6.31" href="#section-6.31">6.31</a> Response-Key</span>

   The Response-Key request-header field can be used by a client to
   request the key that the called user agent SHOULD use to encrypt the
   response with. The syntax is:





<span class="grey">Handley, et al.             Standards Track                    [Page 63]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-64" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Response-Key  =  "Response-Key" ":" key-scheme 1*SP #key-param
        key-scheme    =  token
        key-param     =  token "=" ( token | quoted-string )


   The "key-scheme" gives the type of encryption to be used for the
   response. <a href="#section-13">Section 13</a> describes security schemes.

   If the client insists that the server return an encrypted response,
   it includes a

                  Require: org.ietf.sip.encrypt-response

   header field in its request. If the server cannot encrypt for
   whatever reason, it MUST follow normal Require header field
   procedures and return a 420 (Bad Extension) response. If this Require
   header field is not present, a server SHOULD still encrypt if it can.

<span class="h3"><a class="selflink" id="section-6.32" href="#section-6.32">6.32</a> Retry-After</span>

   The Retry-After general-header field can be used with a 503 (Service
   Unavailable) response to indicate how long the service is expected to
   be unavailable to the requesting client and with a 404 (Not Found),
   600 (Busy), or 603 (Decline) response to indicate when the called
   party anticipates being available again. The value of this field can
   be either an SIP-date or an integer number of seconds (in decimal)
   after the time of the response.

   A REGISTER request MAY include this header field when deleting
   registrations with "Contact: * ;expires: 0". The Retry-After value
   then indicates when the user might again be reachable. The registrar
   MAY then include this information in responses to future calls.

   An optional comment can be used to indicate additional information
   about the time of callback. An optional "duration" parameter
   indicates how long the called party will be reachable starting at the
   initial time of availability. If no duration parameter is given, the
   service is assumed to be available indefinitely.



        Retry-After  =  "Retry-After" ":" ( SIP-date | delta-seconds )
                        [ comment ] [ ";" "duration" "=" delta-seconds ]


   Examples of its use are

     Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)



<span class="grey">Handley, et al.             Standards Track                    [Page 64]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-65" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


     Retry-After: Mon, 01 Jan 9999 00:00:00 GMT
       (Dear John: Don't call me back, ever)
     Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
     Retry-After: 120



   In the third example, the callee is reachable for one hour starting
   at 21:00 GMT. In the last example, the delay is 2 minutes.

<span class="h3"><a class="selflink" id="section-6.33" href="#section-6.33">6.33</a> Route</span>

   The Route request-header field determines the route taken by a
   request. Each host removes the first entry and then proxies the
   request to the host listed in that entry, also using it as the
   Request-URI. The operation is further described in <a href="#section-6.29">Section 6.29</a>.

   The Route header field has the following syntax:


        Route  =  "Route" ":" 1# name-addr


<span class="h3"><a class="selflink" id="section-6.34" href="#section-6.34">6.34</a> Server</span>

   The Server response-header field contains information about the
   software used by the user agent server to handle the request. The
   syntax for this field is defined in [H14.39].

<span class="h3"><a class="selflink" id="section-6.35" href="#section-6.35">6.35</a> Subject</span>

   This is intended to provide a summary, or to indicate the nature, of
   the call, allowing call filtering without having to parse the session
   description. (Also, the session description does not have to use the
   same subject indication as the invitation.)



        Subject  =  ( "Subject" | "s" ) ":" *TEXT-UTF8


   Example:


     Subject: Tune in - they are talking about your work!






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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-66" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.36" href="#section-6.36">6.36</a> Timestamp</span>

   The timestamp general-header field describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and it MAY use any timescale. The server MUST echo
   the exact same value and MAY, if it has accurate information about
   this, add a floating point number indicating the number of seconds
   that have elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions.



        Timestamp  =  "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
        delay      =  *(DIGIT) [ "." *(DIGIT) ]


   Note that there MUST NOT be any LWS between a DIGIT and the decimal
   point.

<span class="h3"><a class="selflink" id="section-6.37" href="#section-6.37">6.37</a> To</span>

   The To general-header field specifies recipient of the request, with
   the same SIP URL syntax as the From field.



        To  =  ( "To" | "t" ) ":" ( name-addr | addr-spec )
               *( ";" addr-params )


   Requests and responses MUST contain a To general-header field,
   indicating the desired recipient of the request. The optional
   "display-name" is meant to be rendered by a human-user interface.
   The UAS or redirect server copies the To header field into its
   response, and MUST add a "tag" parameter if the request contained
   more than one Via header field.


        If there was more than one Via header field, the request
        was handled by at least one proxy server. Since the
        receiver cannot know whether any of the proxy servers
        forked the request, it is safest to assume that they might
        have.

   The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
   "ttl-param", or "headers" elements. A server that receives a SIP-URL
   with these elements removes them before further processing.



<span class="grey">Handley, et al.             Standards Track                    [Page 66]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-67" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The "tag" parameter serves as a general mechanism to distinguish
   multiple instances of a user identified by a single SIP URL. As
   proxies can fork requests, the same request can reach multiple
   instances of a user (mobile and home phones, for example). As each
   can respond, there needs to be a means to distinguish the responses
   from each at the caller. The situation also arises with multicast
   requests. The tag in the To header field serves to distinguish
   responses at the UAC. It MUST be placed in the To field of the
   response by each instance when there is a possibility that the
   request was forked at an intermediate proxy. The "tag" MUST be added
   by UAS, registrars and redirect servers, but MUST NOT be inserted
   into responses forwarded upstream by proxies. The "tag" is added for
   all definitive responses for all methods, and MAY be added for
   informational responses from a UAS or redirect server. All subsequent
   transactions between two entities MUST include the "tag" parameter,
   as described in <a href="#section-11">Section 11</a>.

   See <a href="#section-6.21">Section 6.21</a> for details of the "tag" parameter.

   The "tag" parameter in To headers is ignored when matching responses
   to requests that did not contain a "tag" in their To header.

   A SIP server returns a 400 (Bad Request) response if it receives a
   request with a To header field containing a URI with a scheme it does
   not recognize.

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.

   The following are examples of valid To headers:

     To: The Operator &lt;sip:operator@cs.columbia.edu&gt;;tag=287447
     To: sip:+12125551212@server.phone2net.com




        Call-ID, To and From are needed to identify a call leg.
        The distinction between call and call leg matters in calls
        with multiple responses from a forked request. The "tag" is
        added to the To header field in the response to allow
        forking of future requests for the same call by proxies,
        while addressing only one of the possibly several
        responding user agent servers. It also allows several
        instances of the callee to send requests that can be
        distinguished.




<span class="grey">Handley, et al.             Standards Track                    [Page 67]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-68" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-6.38" href="#section-6.38">6.38</a> Unsupported</span>

   The Unsupported response-header field lists the features not
   supported by the server. See <a href="#section-6.30">Section 6.30</a> for a usage example and
   motivation.

   Syntax:



        Unsupported  =  "Unsupported" ":" 1#option-tag


<span class="h3"><a class="selflink" id="section-6.39" href="#section-6.39">6.39</a> User-Agent</span>

   The User-Agent general-header field contains information about the
   client user agent originating the request. The syntax and semantics
   are defined in [H14.42].

<span class="h3"><a class="selflink" id="section-6.40" href="#section-6.40">6.40</a> Via</span>

   The Via field indicates the path taken by the request so far.  This
   prevents request looping and ensures replies take the same path as
   the requests, which assists in firewall traversal and other unusual
   routing situations.

<span class="h4"><a class="selflink" id="section-6.40.1" href="#section-6.40.1">6.40.1</a> Requests</span>

   The client originating the request MUST insert into the request a Via
   field containing its host name or network address and, if not the
   default port number, the port number at which it wishes to receive
   responses. (Note that this port number can differ from the UDP source
   port number of the request.) A fully-qualified domain name is
   RECOMMENDED. Each subsequent proxy server that sends the request
   onwards MUST add its own additional Via field before any existing Via
   fields. A proxy that receives a redirection (3xx) response and then
   searches recursively, MUST use the same Via headers as on the
   original proxied request.

   A proxy SHOULD check the top-most Via header field to ensure that it
   contains the sender's correct network address, as seen from that
   proxy. If the sender's address is incorrect, the proxy MUST add an
   additional "received" attribute, as described 6.40.2.


        A host behind a network address translator (NAT) or
        firewall may not be able to insert a network address into
        the Via header that can be reached by the next hop beyond



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-69" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        the NAT. Use of the received attribute allows SIP requests
        to traverse NAT's which only modify the source IP address.
        NAT's which modify port numbers, called Network Address
        Port Translator's (NAPT) will not properly pass SIP when
        transported on UDP, in which case an application layer
        gateway is required. When run over TCP, SIP stands a better
        chance of traversing NAT's, since its behavior is similar
        to HTTP in this case (but of course on different ports).

   A proxy sending a request to a multicast address MUST add the "maddr"
   parameter to its Via header field, and SHOULD add the "ttl"
   parameter. If a server receives a request which contained an "maddr"
   parameter in the topmost Via field, it SHOULD send the response to
   the multicast address listed in the "maddr" parameter.

   If a proxy server receives a request which contains its own address
   in the Via header value, it MUST respond with a 482 (Loop Detected)
   status code.

   A proxy server MUST NOT forward a request to a multicast group which
   already appears in any of the Via headers.


        This prevents a malfunctioning proxy server from causing
        loops. Also, it cannot be guaranteed that a proxy server
        can always detect that the address returned by a location
        service refers to a host listed in the Via list, as a
        single host may have aliases or several network interfaces.

<span class="h4"><a class="selflink" id="section-6.40.2" href="#section-6.40.2">6.40.2</a> Receiver-tagged Via Header Fields</span>

   Normally, every host that sends or forwards a SIP message adds a Via
   field indicating the path traversed. However, it is possible that
   Network Address Translators (NATs) changes the source address and
   port of the request (e.g., from net-10 to a globally routable
   address), in which case the Via header field cannot be relied on to
   route replies. To prevent this, a proxy SHOULD check the top-most Via
   header field to ensure that it contains the sender's correct network
   address, as seen from that proxy. If the sender's address is
   incorrect, the proxy MUST add a "received" parameter to the Via
   header field inserted by the previous hop. Such a modified Via header
   field is known as a receiver-tagged Via header field. An example is:


     Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
     Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3





<span class="grey">Handley, et al.             Standards Track                    [Page 69]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-70" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   In this example, the message originated from 10.0.0.1 and traversed a
   NAT with the external address border.ieee.org (199.172.136.3) to
   reach erlang.bell-telephone.com.  The latter noticed the mismatch,
   and added a parameter to the previous hop's Via header field,
   containing the address that the packet actually came from. (Note that
   the NAT border.ieee.org is not a SIP server.)

<span class="h4"><a class="selflink" id="section-6.40.3" href="#section-6.40.3">6.40.3</a> Responses</span>

   Via header fields in responses are processed by a proxy or UAC
   according to the following rules:

        1.   The first Via header field should indicate the proxy or
             client processing this response. If it does not, discard
             the message.  Otherwise, remove this Via field.

        2.   If there is no second Via header field, this response is
             destined for this client. Otherwise, the processing depends
             on whether the Via field contains a "maddr" parameter or is
             a receiver-tagged field:

             - If the second Via header field contains a "maddr"
               parameter, send the response to the multicast address
               listed there, using the port indicated in "sent-by", or
               port 5060 if none is present. The response SHOULD be sent
               using the TTL indicated in the "ttl" parameter, or with a
               TTL of 1 if that parameter is not present. For
               robustness, responses MUST be sent to the address
               indicated in the "maddr" parameter even if it is not a
               multicast address.

             - If the second Via header field does not contain a "maddr"
               parameter and is a receiver-tagged field (<a href="#section-6.40.2">Section</a>
               <a href="#section-6.40.2">6.40.2</a>), send the message to the address in the
               "received" parameter, using the port indicated in the
               "sent-by" value, or using port 5060 if none is present.

             - If neither of the previous cases apply, send the message
               to the address indicated by the "sent-by" value in the
               second Via header field.

<span class="h4"><a class="selflink" id="section-6.40.4" href="#section-6.40.4">6.40.4</a> User Agent and Redirect Servers</span>

   A UAS or redirect server sends a response based on one of the
   following rules:

        o  If the first Via header field in the request contains a
          "maddr" parameter, send the response to the multicast address



<span class="grey">Handley, et al.             Standards Track                    [Page 70]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-71" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


          listed there, using the port indicated in "sent-by", or port
          5060 if none is present. The response SHOULD be sent using the
          TTL indicated in the "ttl" parameter, or with a TTL of 1 if
          that parameter is not present. For robustness, responses MUST
          be sent to the address indicated in the "maddr" parameter even
          if it is not a multicast address.

        o  If the address in the "sent-by" value of the first Via field
          differs from the source address of the packet, send the
          response to the actual packet source address, similar to the
          treatment for receiver-tagged Via header fields (<a href="#section-6.40.2">Section</a>
          <a href="#section-6.40.2">6.40.2</a>).

        o  If neither of these conditions is true, send the response to
          the address contained in the "sent-by" value. If the request
          was sent using TCP, use the existing TCP connection if
          available.

<span class="h4"><a class="selflink" id="section-6.40.5" href="#section-6.40.5">6.40.5</a> Syntax</span>

   The format for a Via header field is shown in Fig. 11. The defaults
   for "protocol-name" and "transport" are "SIP" and "UDP",
   respectively. The "maddr" parameter, designating the multicast
   address, and the "ttl" parameter, designating the time-to-live (TTL)
   value, are included only if the request was sent via multicast. The
   "received" parameter is added only for receiver-added Via fields
   (<a href="#section-6.40.2">Section 6.40.2</a>). For reasons of privacy, a client or proxy may wish
   to hide its Via information by encrypting it (see <a href="#section-6.22">Section 6.22</a>). The
   "hidden" parameter is included if this header field was hidden by the
   upstream proxy (see 6.22). Note that privacy of the proxy relies on
   the cooperation of the next hop, as the next-hop proxy will, by
   necessity, know the IP address and port number of the source host.


   The "branch" parameter is included by every forking proxy.  The token
   MUST be unique for each distinct request generated when a proxy
   forks. CANCEL requests MUST have the same branch value as the
   corresponding forked request. When a response arrives at the proxy it
   can use the branch value to figure out which branch the response
   corresponds to. A proxy which generates a single request (non-
   forking) MAY also insert the "branch" parameter. The identifier has
   to be unique only within a set of isomorphic requests.


     Via: SIP/2.0/UDP first.example.com:4000;ttl=16
       ;maddr=224.2.0.1 ;branch=a7c6a8dlze (Example)
     Via: SIP/2.0/UDP adk8




<span class="grey">Handley, et al.             Standards Track                    [Page 71]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-72" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




  Via              = ( "Via" | "v") ":" 1#( sent-protocol sent-by
                     *( ";" via-params ) [ comment ] )
  via-params       = via-hidden | via-ttl | via-maddr
                   | via-received | via-branch
  via-hidden       = "hidden"
  via-ttl          = "ttl" "=" ttl
  via-maddr        = "maddr" "=" maddr
  via-received     = "received" "=" host
  via-branch       = "branch" "=" token
  sent-protocol    = protocol-name "/" protocol-version "/" transport
  protocol-name    = "SIP" | token
  protocol-version = token
  transport        = "UDP" | "TCP" | token
  sent-by          = ( host [ ":" port ] ) | ( concealed-host )
  concealed-host   = token
  ttl              = 1*3DIGIT     ; 0 to 255


   Figure 11: Syntax of Via header field


<span class="h3"><a class="selflink" id="section-6.41" href="#section-6.41">6.41</a> Warning</span>

   The Warning response-header field is used to carry additional
   information about the status of a response. Warning headers are sent
   with responses and have the following format:



        Warning        =  "Warning" ":" 1#warning-value
        warning-value  =  warn-code SP warn-agent SP warn-text
        warn-code      =  3DIGIT
        warn-agent     =  ( host [ ":" port ] ) | pseudonym
                          ;  the name or pseudonym of the server adding
                          ;  the Warning header, for use in debugging
        warn-text      =  quoted-string


   A response MAY carry more than one Warning header.

   The "warn-text" should be in a natural language that is most likely
   to be intelligible to the human user receiving the response.  This
   decision can be based on any available knowledge, such as the
   location of the cache or user, the Accept-Language field in a
   request, or the Content-Language field in a response. The default
   language is i-default [<a href="#ref-31" title="&quot;IETF policy on character sets and languages&quot;">31</a>].



<span class="grey">Handley, et al.             Standards Track                    [Page 72]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-73" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Any server MAY add Warning headers to a response. Proxy servers MUST
   place additional Warning headers before any Authorization headers.
   Within that constraint, Warning headers MUST be added after any
   existing Warning headers not covered by a signature. A proxy server
   MUST NOT delete any Warning header field that it received with a
   response.

   When multiple Warning headers are attached to a response, the user
   agent SHOULD display as many of them as possible, in the order that
   they appear in the response. If it is not possible to display all of
   the warnings, the user agent first displays warnings that appear
   early in the response.

   The warn-code consists of three digits. A first digit of "3"
   indicates warnings specific to SIP.

   This is a list of the currently-defined "warn-code"s, each with a
   recommended warn-text in English, and a description of its meaning.
   Note that these warnings describe failures induced by the session
   description.

   Warnings 300 through 329 are reserved for indicating problems with
   keywords in the session description, 330 through 339 are warnings
   related to basic network services requested in the session
   description, 370 through 379 are warnings related to quantitative QoS
   parameters requested in the session description, and 390 through 399
   are miscellaneous warnings that do not fall into one of the above
   categories.

   300 Incompatible network protocol: One or more network protocols
        contained in the session description are not available.

   301 Incompatible network address formats: One or more network address
        formats contained in the session description are not available.

   302 Incompatible transport protocol: One or more transport protocols
        described in the session description are not available.

   303 Incompatible bandwidth units: One or more bandwidth measurement
        units contained in the session description were not understood.

   304 Media type not available: One or more media types contained in
        the session description are not available.

   305 Incompatible media format: One or more media formats contained in
        the session description are not available.





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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   306 Attribute not understood: One or more of the media attributes in
        the session description are not supported.

   307 Session description parameter not understood: A parameter other
        than those listed above was not understood.

   330 Multicast not available: The site where the user is located does
        not support multicast.

   331 Unicast not available: The site where the user is located does
        not support unicast communication (usually due to the presence
        of a firewall).

   370 Insufficient bandwidth: The bandwidth specified in the session
        description or defined by the media exceeds that known to be
        available.

   399 Miscellaneous warning: The warning text can include arbitrary
        information to be presented to a human user, or logged. A system
        receiving this warning MUST NOT take any automated action.


        1xx and 2xx have been taken by HTTP/1.1.

   Additional "warn-code"s, as in the example below, can be defined
   through IANA.

   Examples:


     Warning: 307 isi.edu "Session parameter 'foo' not understood"
     Warning: 301 isi.edu "Incompatible network address type 'E.164'"



<span class="h3"><a class="selflink" id="section-6.42" href="#section-6.42">6.42</a> WWW-Authenticate</span>

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages. The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI. See [H14.46] for a
   definition of the syntax, and <a href="#section-14">section 14</a> for an overview of usage.

   The content of the "realm" parameter SHOULD be displayed to the user.
   A user agent SHOULD cache the authorization credentials for a given
   value of the destination (To header) and "realm" and attempt to re-
   use these values on the next request for that destination.




<span class="grey">Handley, et al.             Standards Track                    [Page 74]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-75" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   In addition to the "basic" and "digest" authentication schemes
   defined in the specifications cited above, SIP defines a new scheme,
   PGP (<a href="./rfc2015">RFC 2015</a>, [<a href="#ref-32" title="&quot;MIME security with pretty good privacy (PGP)&quot;">32</a>]), <a href="#section-15">Section 15</a>. Other schemes, such as S/MIME, are
   for further study.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a> Status Code Definitions</span>

   The response codes are consistent with, and extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate, and only
   those that are appropriate are given here. Other HTTP/1.1 response
   codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have
   codes x80 upwards to avoid clashes with future HTTP response codes.
   Also, SIP defines a new class, 6xx. The default behavior for unknown
   response codes is given for each category of codes.

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a> Informational 1xx</span>

   Informational responses indicate that the server or proxy contacted
   is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from the
   server, and the server SHOULD send such a response without further
   prompting. A server SHOULD send a 1xx response if it expects to take
   more than 200 ms to obtain a final response. A server MAY issue zero
   or more 1xx responses, with no restriction on their ordering or
   uniqueness. Note that 1xx responses are not transmitted reliably,
   that is, they do not cause the client to send an ACK. Servers are
   free to retransmit informational responses and clients can inquire
   about the current state of call processing by re-sending the request.

<span class="h4"><a class="selflink" id="section-7.1.1" href="#section-7.1.1">7.1.1</a> 100 Trying</span>

   Some unspecified action is being taken on behalf of this call (e.g.,
   a database is being consulted), but the user has not yet been
   located.

<span class="h4"><a class="selflink" id="section-7.1.2" href="#section-7.1.2">7.1.2</a> 180 Ringing</span>

   The called user agent has located a possible location where the user
   has registered recently and is trying to alert the user.

<span class="h4"><a class="selflink" id="section-7.1.3" href="#section-7.1.3">7.1.3</a> 181 Call Is Being Forwarded</span>

   A proxy server MAY use this status code to indicate that the call is
   being forwarded to a different set of destinations.







<span class="grey">Handley, et al.             Standards Track                    [Page 75]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-76" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.1.4" href="#section-7.1.4">7.1.4</a> 182 Queued</span>

   The called party is temporarily unavailable, but the callee has
   decided to queue the call rather than reject it. When the callee
   becomes available, it will return the appropriate final status
   response. The reason phrase MAY give further details about the status
   of the call, e.g., "5 calls queued; expected waiting time is 15
   minutes". The server MAY issue several 182 responses to update the
   caller about the status of the queued call.

<span class="h3"><a class="selflink" id="section-7.2" href="#section-7.2">7.2</a> Successful 2xx</span>

   The request was successful and MUST terminate a search.

<span class="h4"><a class="selflink" id="section-7.2.1" href="#section-7.2.1">7.2.1</a> 200 OK</span>

   The request has succeeded. The information returned with the response
   depends on the method used in the request, for example:

   BYE: The call has been terminated. The message body is empty.

   CANCEL: The search has been cancelled. The message body is empty.

   INVITE: The callee has agreed to participate; the message body
        indicates the callee's capabilities.

   OPTIONS: The callee has agreed to share its capabilities, included in
        the message body.

   REGISTER: The registration has succeeded. The client treats the
        message body according to its Content-Type.

<span class="h3"><a class="selflink" id="section-7.3" href="#section-7.3">7.3</a> Redirection 3xx</span>

   3xx responses give information about the user's new location, or
   about alternative services that might be able to satisfy the call.
   They SHOULD terminate an existing search, and MAY cause the initiator
   to begin a new search if appropriate.

   Any redirection (3xx) response MUST NOT suggest any of the addresses
   in the Via (<a href="#section-6.40">Section 6.40</a>) path of the request in the Contact header
   field. (Addresses match if their host and port number match.)

   To avoid forwarding loops, a user agent client or proxy MUST check
   whether the address returned by a redirect server equals an address
   tried earlier.





<span class="grey">Handley, et al.             Standards Track                    [Page 76]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-77" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.3.1" href="#section-7.3.1">7.3.1</a> 300 Multiple Choices</span>

   The address in the request resolved to several choices, each with its
   own specific location, and the user (or user agent) can select a
   preferred communication end point and redirect its request to that
   location.

   The response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the user or user agent can
   choose the one most appropriate, if allowed by the Accept request
   header. The entity format is specified by the media type given in the
   Content-Type header field. The choices SHOULD also be listed as
   Contact fields (<a href="#section-6.13">Section 6.13</a>).  Unlike HTTP, the SIP response MAY
   contain several Contact fields or a list of addresses in a Contact
   field. User agents MAY use the Contact header field value for
   automatic redirection or MAY ask the user to confirm a choice.
   However, this specification does not define any standard for such
   automatic selection.


        This status response is appropriate if the callee can be
        reached at several different locations and the server
        cannot or prefers not to proxy the request.

<span class="h4"><a class="selflink" id="section-7.3.2" href="#section-7.3.2">7.3.2</a> 301 Moved Permanently</span>

   The user can no longer be found at the address in the Request-URI and
   the requesting client SHOULD retry at the new address given by the
   Contact header field (<a href="#section-6.13">Section 6.13</a>). The caller SHOULD update any
   local directories, address books and user location caches with this
   new value and redirect future requests to the address(es) listed.

<span class="h4"><a class="selflink" id="section-7.3.3" href="#section-7.3.3">7.3.3</a> 302 Moved Temporarily</span>

   The requesting client SHOULD retry the request at the new address(es)
   given by the Contact header field (<a href="#section-6.13">Section 6.13</a>).  The duration of
   the redirection can be indicated through an Expires (<a href="#section-6.20">Section 6.20</a>)
   header. If there is no explicit expiration time, the address is only
   valid for this call and MUST NOT be cached for future calls.

<span class="h4"><a class="selflink" id="section-7.3.4" href="#section-7.3.4">7.3.4</a> 305 Use Proxy</span>

   The requested resource MUST be accessed through the proxy given by
   the Contact field. The Contact field gives the URI of the proxy. The
   recipient is expected to repeat this single request via the proxy.
   305 responses MUST only be generated by user agent servers.





<span class="grey">Handley, et al.             Standards Track                    [Page 77]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-78" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.3.5" href="#section-7.3.5">7.3.5</a> 380 Alternative Service</span>

   The call was not successful, but alternative services are possible.
   The alternative services are described in the message body of the
   response.  Formats for such bodies are not defined here, and may be
   the subject of future standardization.

<span class="h3"><a class="selflink" id="section-7.4" href="#section-7.4">7.4</a> Request Failure 4xx</span>

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (e.g., adding appropriate authorization). However, the
   same request to a different server might be successful.

<span class="h4"><a class="selflink" id="section-7.4.1" href="#section-7.4.1">7.4.1</a> 400 Bad Request</span>

   The request could not be understood due to malformed syntax.

<span class="h4"><a class="selflink" id="section-7.4.2" href="#section-7.4.2">7.4.2</a> 401 Unauthorized</span>

   The request requires user authentication.

<span class="h4"><a class="selflink" id="section-7.4.3" href="#section-7.4.3">7.4.3</a> 402 Payment Required</span>

   Reserved for future use.

<span class="h4"><a class="selflink" id="section-7.4.4" href="#section-7.4.4">7.4.4</a> 403 Forbidden</span>

   The server understood the request, but is refusing to fulfill it.
   Authorization will not help, and the request SHOULD NOT be repeated.

<span class="h4"><a class="selflink" id="section-7.4.5" href="#section-7.4.5">7.4.5</a> 404 Not Found</span>

   The server has definitive information that the user does not exist at
   the domain specified in the Request-URI. This status is also returned
   if the domain in the Request-URI does not match any of the domains
   handled by the recipient of the request.

<span class="h4"><a class="selflink" id="section-7.4.6" href="#section-7.4.6">7.4.6</a> 405 Method Not Allowed</span>

   The method specified in the Request-Line is not allowed for the
   address identified by the Request-URI. The response MUST include an
   Allow header field containing a list of valid methods for the
   indicated address.







<span class="grey">Handley, et al.             Standards Track                    [Page 78]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-79" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.4.7" href="#section-7.4.7">7.4.7</a> 406 Not Acceptable</span>

   The resource identified by the request is only capable of generating
   response entities which have content characteristics not acceptable
   according to the accept headers sent in the request.

<span class="h4"><a class="selflink" id="section-7.4.8" href="#section-7.4.8">7.4.8</a> 407 Proxy Authentication Required</span>

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. The proxy MUST
   return a Proxy-Authenticate header field (<a href="#section-6.26">section 6.26</a>) containing a
   challenge applicable to the proxy for the requested resource. The
   client MAY repeat the request with a suitable Proxy-Authorization
   header field (<a href="#section-6.27">section 6.27</a>). SIP access authentication is explained
   in <a href="#section-13.2">section 13.2</a> and 14.

   This status code is used for applications where access to the
   communication channel (e.g., a telephony gateway) rather than the
   callee requires authentication.

<span class="h4"><a class="selflink" id="section-7.4.9" href="#section-7.4.9">7.4.9</a> 408 Request Timeout</span>

   The server could not produce a response, e.g., a user location,
   within the time indicated in the Expires request-header field. The
   client MAY repeat the request without modifications at any later
   time.

<span class="h4"><a class="selflink" id="section-7.4.10" href="#section-7.4.10">7.4.10</a> 409 Conflict</span>

   The request could not be completed due to a conflict with the current
   state of the resource. This response is returned if the action
   parameter in a REGISTER request conflicts with existing
   registrations.

<span class="h4"><a class="selflink" id="section-7.4.11" href="#section-7.4.11">7.4.11</a> 410 Gone</span>

   The requested resource is no longer available at the server and no
   forwarding address is known. This condition is expected to be
   considered permanent. If the server does not know, or has no facility
   to determine, whether or not the condition is permanent, the status
   code 404 (Not Found) SHOULD be used instead.

<span class="h4"><a class="selflink" id="section-7.4.12" href="#section-7.4.12">7.4.12</a> 411 Length Required</span>

   The server refuses to accept the request without a defined Content-
   Length. The client MAY repeat the request if it adds a valid
   Content-Length header field containing the length of the message-body
   in the request message.



<span class="grey">Handley, et al.             Standards Track                    [Page 79]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-80" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.4.13" href="#section-7.4.13">7.4.13</a> 413 Request Entity Too Large</span>

   The server is refusing to process a request because the request
   entity is larger than the server is willing or able to process. The
   server MAY close the connection to prevent the client from continuing
   the request.

   If the condition is temporary, the server SHOULD include a Retry-
   After header field to indicate that it is temporary and after what
   time the client MAY try again.

<span class="h4"><a class="selflink" id="section-7.4.14" href="#section-7.4.14">7.4.14</a> 414 Request-URI Too Long</span>

   The server is refusing to service the request because the Request-URI
   is longer than the server is willing to interpret.

<span class="h4"><a class="selflink" id="section-7.4.15" href="#section-7.4.15">7.4.15</a> 415 Unsupported Media Type</span>

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the requested
   resource for the requested method. The server SHOULD return a list of
   acceptable formats using the Accept, Accept-Encoding and Accept-
   Language header fields.

<span class="h4"><a class="selflink" id="section-7.4.16" href="#section-7.4.16">7.4.16</a> 420 Bad Extension</span>

   The server did not understand the protocol extension specified in a
   Require (<a href="#section-6.30">Section 6.30</a>) header field.

<span class="h4"><a class="selflink" id="section-7.4.17" href="#section-7.4.17">7.4.17</a> 480 Temporarily Unavailable</span>

   The callee's end system was contacted successfully but the callee is
   currently unavailable (e.g., not logged in or logged in in such a
   manner as to preclude communication with the callee). The response
   MAY indicate a better time to call in the Retry-After header. The
   user could also be available elsewhere (unbeknownst to this host),
   thus, this response does not terminate any searches. The reason
   phrase SHOULD indicate a more precise cause as to why the callee is
   unavailable. This value SHOULD be setable by the user agent. Status
   486 (Busy Here) MAY be used to more precisely indicate a particular
   reason for the call failure.

   This status is also returned by a redirect server that recognizes the
   user identified by the Request-URI, but does not currently have a
   valid forwarding location for that user.






<span class="grey">Handley, et al.             Standards Track                    [Page 80]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-81" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.4.18" href="#section-7.4.18">7.4.18</a> 481 Call Leg/Transaction Does Not Exist</span>

   This status is returned under two conditions: The server received a
   BYE request that does not match any existing call leg or the server
   received a CANCEL request that does not match any existing
   transaction. (A server simply discards an ACK referring to an unknown
   transaction.)

<span class="h4"><a class="selflink" id="section-7.4.19" href="#section-7.4.19">7.4.19</a> 482 Loop Detected</span>

   The server received a request with a Via (<a href="#section-6.40">Section 6.40</a>) path
   containing itself.

<span class="h4"><a class="selflink" id="section-7.4.20" href="#section-7.4.20">7.4.20</a> 483 Too Many Hops</span>

   The server received a request that contains more Via entries (hops)
   (<a href="#section-6.40">Section 6.40</a>) than allowed by the Max-Forwards (<a href="#section-6.23">Section 6.23</a>) header
   field.

<span class="h4"><a class="selflink" id="section-7.4.21" href="#section-7.4.21">7.4.21</a> 484 Address Incomplete</span>

   The server received a request with a To (<a href="#section-6.37">Section 6.37</a>) address or
   Request-URI that was incomplete. Additional information SHOULD be
   provided.


        This status code allows overlapped dialing. With overlapped
        dialing, the client does not know the length of the dialing
        string. It sends strings of increasing lengths, prompting
        the user for more input, until it no longer receives a 484
        status response.

<span class="h4"><a class="selflink" id="section-7.4.22" href="#section-7.4.22">7.4.22</a> 485 Ambiguous</span>

   The callee address provided in the request was ambiguous. The
   response MAY contain a listing of possible unambiguous addresses in
   Contact headers.

   Revealing alternatives can infringe on privacy concerns of the user
   or the organization. It MUST be possible to configure a server to
   respond with status 404 (Not Found) or to suppress the listing of
   possible choices if the request address was ambiguous.

   Example response to a request with the URL lee@example.com :

   485 Ambiguous SIP/2.0
   Contact: Carol Lee &lt;sip:carol.lee@example.com&gt;




<span class="grey">Handley, et al.             Standards Track                    [Page 81]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-82" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Contact: Ping Lee &lt;sip:p.lee@example.com&gt;
   Contact: Lee M. Foote &lt;sip:lee.foote@example.com&gt;



        Some email and voice mail systems provide this
        functionality. A status code separate from 3xx is used
        since the semantics are different: for 300, it is assumed
        that the same person or service will be reached by the
        choices provided. While an automated choice or sequential
        search makes sense for a 3xx response, user intervention is
        required for a 485 response.

<span class="h4"><a class="selflink" id="section-7.4.23" href="#section-7.4.23">7.4.23</a> 486 Busy Here</span>

   The callee's end system was contacted successfully but the callee is
   currently not willing or able to take additional calls. The response
   MAY indicate a better time to call in the Retry-After header. The
   user could also be available elsewhere, such as through a voice mail
   service, thus, this response does not terminate any searches.  Status
   600 (Busy Everywhere) SHOULD be used if the client knows that no
   other end system will be able to accept this call.

<span class="h3"><a class="selflink" id="section-7.5" href="#section-7.5">7.5</a> Server Failure 5xx</span>

   5xx responses are failure responses given when a server itself has
   erred. They are not definitive failures, and MUST NOT terminate a
   search if other possible locations remain untried.

<span class="h4"><a class="selflink" id="section-7.5.1" href="#section-7.5.1">7.5.1</a> 500 Server Internal Error</span>

   The server encountered an unexpected condition that prevented it from
   fulfilling the request. The client MAY display the specific error
   condition, and MAY retry the request after several seconds.

<span class="h4"><a class="selflink" id="section-7.5.2" href="#section-7.5.2">7.5.2</a> 501 Not Implemented</span>

   The server does not support the functionality required to fulfill the
   request. This is the appropriate response when the server does not
   recognize the request method and is not capable of supporting it for
   any user.

<span class="h4"><a class="selflink" id="section-7.5.3" href="#section-7.5.3">7.5.3</a> 502 Bad Gateway</span>

   The server, while acting as a gateway or proxy, received an invalid
   response from the downstream server it accessed in attempting to
   fulfill the request.




<span class="grey">Handley, et al.             Standards Track                    [Page 82]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-83" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.5.4" href="#section-7.5.4">7.5.4</a> 503 Service Unavailable</span>

   The server is currently unable to handle the request due to a
   temporary overloading or maintenance of the server. The implication
   is that this is a temporary condition which will be alleviated after
   some delay. If known, the length of the delay MAY be indicated in a
   Retry-After header. If no Retry-After is given, the client MUST
   handle the response as it would for a 500 response.

   Note: The existence of the 503 status code does not imply that a
   server has to use it when becoming overloaded. Some servers MAY wish
   to simply refuse the connection.

<span class="h4"><a class="selflink" id="section-7.5.5" href="#section-7.5.5">7.5.5</a> 504 Gateway Time-out</span>

   The server, while acting as a gateway, did not receive a timely
   response from the server (e.g., a location server) it accessed in
   attempting to complete the request.

<span class="h4"><a class="selflink" id="section-7.5.6" href="#section-7.5.6">7.5.6</a> 505 Version Not Supported</span>

   The server does not support, or refuses to support, the SIP protocol
   version that was used in the request message. The server is
   indicating that it is unable or unwilling to complete the request
   using the same major version as the client, other than with this
   error message. The response MAY contain an entity describing why that
   version is not supported and what other protocols are supported by
   that server. The format for such an entity is not defined here and
   may be the subject of future standardization.

<span class="h3"><a class="selflink" id="section-7.6" href="#section-7.6">7.6</a> Global Failures 6xx</span>

   6xx responses indicate that a server has definitive information about
   a particular user, not just the particular instance indicated in the
   Request-URI. All further searches for this user are doomed to failure
   and pending searches SHOULD be terminated.

<span class="h4"><a class="selflink" id="section-7.6.1" href="#section-7.6.1">7.6.1</a> 600 Busy Everywhere</span>

   The callee's end system was contacted successfully but the callee is
   busy and does not wish to take the call at this time. The response
   MAY indicate a better time to call in the Retry-After header. If the
   callee does not wish to reveal the reason for declining the call, the
   callee uses status code 603 (Decline) instead. This status response
   is returned only if the client knows that no other end point (such as
   a voice mail system) will answer the request. Otherwise, 486 (Busy
   Here) should be returned.




<span class="grey">Handley, et al.             Standards Track                    [Page 83]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-84" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-7.6.2" href="#section-7.6.2">7.6.2</a> 603 Decline</span>

   The callee's machine was successfully contacted but the user
   explicitly does not wish to or cannot participate. The response MAY
   indicate a better time to call in the Retry-After header.

<span class="h4"><a class="selflink" id="section-7.6.3" href="#section-7.6.3">7.6.3</a> 604 Does Not Exist Anywhere</span>

   The server has authoritative information that the user indicated in
   the To request field does not exist anywhere. Searching for the user
   elsewhere will not yield any results.

<span class="h4"><a class="selflink" id="section-7.6.4" href="#section-7.6.4">7.6.4</a> 606 Not Acceptable</span>

   The user's agent was contacted successfully but some aspects of the
   session description such as the requested media, bandwidth, or
   addressing style were not acceptable.

   A 606 (Not Acceptable) response means that the user wishes to
   communicate, but cannot adequately support the session described. The
   606 (Not Acceptable) response MAY contain a list of reasons in a
   Warning header field describing why the session described cannot be
   supported. Reasons are listed in <a href="#section-6.41">Section 6.41</a>.  It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join an already existing conference, negotiation may
   not be possible. It is up to the invitation initiator to decide
   whether or not to act on a 606 (Not Acceptable) response.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a> SIP Message Body</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a> Body Inclusion</span>

   Requests MAY contain message bodies unless otherwise noted. Within
   this specification, the BYE request MUST NOT contain a message body.
   For ACK, INVITE and OPTIONS, the message body is always a session
   description. The use of message bodies for REGISTER requests is for
   further study.

   For response messages, the request method and the response status
   code determine the type and interpretation of any message body. All
   responses MAY include a body. Message bodies for 1xx responses
   contain advisory information about the progress of the request. 2xx
   responses to INVITE requests contain session descriptions. In 3xx
   responses, the message body MAY contain the description of
   alternative destinations or services, as described in <a href="#section-7.3">Section 7.3</a>.
   For responses with status 400 or greater, the message body MAY





<span class="grey">Handley, et al.             Standards Track                    [Page 84]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-85" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   contain additional, human-readable information about the reasons for
   failure. It is RECOMMENDED that information in 1xx and 300 and
   greater responses be of type text/plain or text/html

<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a> Message Body Type</span>

   The Internet media type of the message body MUST be given by the
   Content-Type header field. If the body has undergone any encoding
   (such as compression) then this MUST be indicated by the Content-
   Encoding header field, otherwise Content-Encoding MUST be omitted. If
   applicable, the character set of the message body is indicated as
   part of the Content-Type header-field value.

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a> Message Body Length</span>

   The body length in bytes SHOULD be given by the Content-Length header
   field. <a href="#section-6.15">Section 6.15</a> describes the behavior in detail.

   The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   to transfer it as a series of chunks, each with its own size
   indicator.)

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a> Compact Form</span>

   When SIP is carried over UDP with authentication and a complex
   session description, it may be possible that the size of a request or
   response is larger than the MTU. To address this problem, a more
   compact form of SIP is also defined by using abbreviations for the
   common header fields listed below:


   short field name  long field name   note
   c                 Content-Type
   e                 Content-Encoding
   f                 From
   i                 Call-ID
   m                 Contact           from "moved"
   l                 Content-Length
   s                 Subject
   t                 To
   v                 Via


   Thus, the message in <a href="#section-16.2">section 16.2</a> could also be written:






<span class="grey">Handley, et al.             Standards Track                    [Page 85]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-86" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


     INVITE sip:schooler@vlsi.caltech.edu SIP/2.0
     v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
     v:SIP/2.0/UDP 128.16.64.19
     f:sip:mjh@isi.edu
     t:sip:schooler@cs.caltech.edu
     i:62729-27@128.16.64.19
     c:application/sdp
     CSeq: 4711 INVITE
     l:187

     v=0
     o=user1 53655765 2353687637 IN IP4 128.3.4.5
     s=Mbone Audio
     i=Discussion of Mbone Engineering Issues
     e=mbone@somewhere.com
     c=IN IP4 224.2.0.1/127
     t=0 0
     m=audio 3456 RTP/AVP 0



   Clients MAY mix short field names and long field names within the
   same request. Servers MUST accept both short and long field names for
   requests. Proxies MAY change header fields between their long and
   short forms, but this MUST NOT be done to fields following an
   Authorization header.

<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a> Behavior of SIP Clients and Servers</span>

<span class="h3"><a class="selflink" id="section-10.1" href="#section-10.1">10.1</a> General Remarks</span>

   SIP is defined so it can use either UDP (unicast or multicast) or TCP
   as a transport protocol; it provides its own reliability mechanism.

<span class="h4"><a class="selflink" id="section-10.1.1" href="#section-10.1.1">10.1.1</a> Requests</span>

   Servers discard isomorphic requests, but first retransmit the
   appropriate response. (SIP requests are said to be idempotent , i.e.,
   receiving more than one copy of a request does not change the server
   state.)

   After receiving a CANCEL request from an upstream client, a stateful
   proxy server MAY send a CANCEL on all branches where it has not yet
   received a final response.

   When a user agent receives a request, it checks the Call-ID against
   those of in-progress calls. If the Call-ID was found, it compares the
   tag value of To with the user's tag and rejects the request if the



<span class="grey">Handley, et al.             Standards Track                    [Page 86]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-87" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   two do not match. If the From header, including any tag value,
   matches the value for an existing call leg, the server compares the
   CSeq header field value. If less than or equal to the current
   sequence number, the request is a retransmission.  Otherwise, it is a
   new request. If the From header does not match an existing call leg,
   a new call leg is created.

   If the Call-ID was not found, a new call leg is created, with entries
   for the To, From and Call-ID headers.  In this case, the To header
   field should not have contained a tag. The server returns a response
   containing the same To value, but with a unique tag added. The tag
   MAY be omitted if the request contained only one Via header field.

<span class="h4"><a class="selflink" id="section-10.1.2" href="#section-10.1.2">10.1.2</a> Responses</span>

   A server MAY issue one or more provisional responses at any time
   before sending a final response. If a stateful proxy, user agent
   server, redirect server or registrar cannot respond to a request with
   a final response within 200 ms, it SHOULD issue a provisional (1xx)
   response as soon as possible. Stateless proxies MUST NOT issue
   provisional responses on their own.

   Responses are mapped to requests by the matching To, From, Call-ID,
   CSeq headers and the branch parameter of the first Via header.
   Responses terminate request retransmissions even if they have Via
   headers that cause them to be delivered to an upstream client.

   A stateful proxy may receive a response that it does not have state
   for, that is, where it has no a record of an associated request. If
   the Via header field indicates that the upstream server used TCP, the
   proxy actively opens a TCP connection to that address. Thus, proxies
   have to be prepared to receive responses on the incoming side of
   passive TCP connections, even though most responses will arrive on
   the incoming side of an active connection. (An active connection is a
   TCP connection initiated by the proxy, a passive connection is one
   accepted by the proxy, but initiated by another entity.)

   100 responses SHOULD NOT be forwarded, other 1xx responses MAY be
   forwarded, possibly after the server eliminates responses with status
   codes that had already been sent earlier. 2xx responses are forwarded
   according to the Via header. Once a stateful proxy has received a 2xx
   response, it MUST NOT forward non-2xx final responses.  Responses
   with status 300 and higher are retransmitted by each stateful proxy
   until the next upstream proxy sends an ACK (see below for timing
   details) or CANCEL.






<span class="grey">Handley, et al.             Standards Track                    [Page 87]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-88" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   A stateful proxy SHOULD maintain state for at least 32 seconds after
   the receipt of the first definitive non-200 response, in order to
   handle retransmissions of the response.


        The 32 second window is given by the maximum retransmission
        duration of 200-class responses using the default timers,
        in case the ACK is lost somewhere on the way to the called
        user agent or the next stateful proxy.

<span class="h3"><a class="selflink" id="section-10.2" href="#section-10.2">10.2</a> Source Addresses, Destination Addresses and Connections</span>

<span class="h4"><a class="selflink" id="section-10.2.1" href="#section-10.2.1">10.2.1</a> Unicast UDP</span>

   Responses are returned to the address listed in the Via header field
   (<a href="#section-6.40">Section 6.40</a>), not the source address of the request.


        Recall that responses are not generated by the next-hop
        stateless server, but generated by either a proxy server or
        the user agent server. Thus, the stateless proxy can only
        use the Via header field to forward the response.

<span class="h4"><a class="selflink" id="section-10.2.2" href="#section-10.2.2">10.2.2</a> Multicast UDP</span>

   Requests MAY be multicast; multicast requests likely feature a host-
   independent Request-URI. This request SHOULD be scoped to ensure it
   is not forwarded beyond the boundaries of the administrative system.
   This MAY be done with either TTL or administrative scopes[25],
   depending on what is implemented in the network.

   A client receiving a multicast query does not have to check whether
   the host part of the Request-URI matches its own host or domain name.
   If the request was received via multicast, the response is also
   returned via multicast. Responses to multicast requests are multicast
   with the same TTL as the request, where the TTL is derived from the
   ttl parameter in the Via header (<a href="#section-6.40">Section 6.40</a>).

   To avoid response implosion, servers MUST NOT answer multicast
   requests with a status code other than 2xx or 6xx. The server delays
   its response by a random interval uniformly distributed between zero
   and one second. Servers MAY suppress responses if they hear a lower-
   numbered or 6xx response from another group member prior to sending.
   Servers do not respond to CANCEL requests received via multicast to
   avoid request implosion. A proxy or UAC SHOULD send a CANCEL on
   receiving the first 2xx or 6xx response to a multicast request.





<span class="grey">Handley, et al.             Standards Track                    [Page 88]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-89" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Server response suppression is a MAY since it requires a
        server to violate some basic message processing rules. Lets
        say A sends a multicast request, and it is received by B,C,
        and D. B sends a 200 response. The topmost Via field in the
        response will contain the address of A. C will also receive
        this response, and could use it to suppress its own
        response. However, C would normally not examine this
        response, as the topmost Via is not its own. Normally, a
        response received with an incorrect topmost Via MUST be
        dropped, but not in this case. To distinguish this packet
        from a misrouted or multicast looped packet is fairly
        complex, and for this reason the procedure is a MAY. The
        CANCEL, instead, provides a simpler and more standard way
        to perform response suppression. It is for this reason that
        the use of CANCEL here is a SHOULD

<span class="h3"><a class="selflink" id="section-10.3" href="#section-10.3">10.3</a> TCP</span>

   A single TCP connection can serve one or more SIP transactions. A
   transaction contains zero or more provisional responses followed by
   one or more final responses. (Typically, transactions contain exactly
   one final response, but there are exceptional circumstances, where,
   for example, multiple 200 responses can be generated.)

   The client SHOULD keep the connection open at least until the first
   final response arrives. If the client closes or resets the TCP
   connection prior to receiving the first final response, the server
   treats this action as equivalent to a CANCEL request.


        This behavior makes it less likely that malfunctioning
        clients cause a proxy server to keep connection state
        indefinitely.

   The server SHOULD NOT close the TCP connection until it has sent its
   final response, at which point it MAY close the TCP connection if it
   wishes to. However, normally it is the client's responsibility to
   close the connection.

   If the server leaves the connection open, and if the client so
   desires it MAY re-use the connection for further SIP requests or for
   requests from the same family of protocols (such as HTTP or stream
   control commands).








<span class="grey">Handley, et al.             Standards Track                    [Page 89]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-90" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   If a server needs to return a response to a client and no longer has
   a connection open to that client, it MAY open a connection to the
   address listed in the Via header. Thus, a proxy or user agent MUST be
   prepared to receive both requests and responses on a "passive"
   connection.

<span class="h3"><a class="selflink" id="section-10.4" href="#section-10.4">10.4</a> Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests</span>

<span class="h4"><a class="selflink" id="section-10.4.1" href="#section-10.4.1">10.4.1</a> UDP</span>

   A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
   REGISTER request with an exponential backoff, starting at a T1 second
   interval, doubling the interval for each packet, and capping off at a
   T2 second interval. This means that after the first packet is sent,
   the second is sent T1 seconds later, the next 2*T1 seconds after
   that, the next 4*T1 seconds after that, and so on, until the interval
   hits T2. Subsequent retransmissions are spaced by T2 seconds. If the
   client receives a provisional response, it continues to retransmit
   the request, but with an interval of T2 seconds.  Retransmissions
   cease when the client has sent a total of eleven packets, or receives
   a definitive response. Default values for T1 and T2 are 500 ms and 4
   s, respectively. Clients MAY use larger values, but SHOULD NOT use
   smaller ones. Servers retransmit the response upon receipt of a
   request retransmission. After the server sends a final response, it
   cannot be sure the client has received the response, and thus SHOULD
   cache the results for at least 10*T2 seconds to avoid having to, for
   example, contact the user or location server again upon receiving a
   request retransmission.


        Use of the exponential backoff is for congestion control
        purposes. However, the back-off must cap off, since request
        retransmissions are used to trigger response
        retransmissions at the server. Without a cap, the loss of a
        single response could significantly increase transaction
        latencies.

   The value of the initial retransmission timer is smaller than that
   that for TCP since it is expected that network paths suitable for
   interactive communications have round-trip times smaller than 500 ms.
   For congestion control purposes, the retransmission count has to be
   bounded.  Given that most transactions are expected to consist of one
   request and a few responses, round-trip time estimation is not likely
   to be very useful. If RTT estimation is desired to more quickly
   discover a missing final response, each request retransmission needs
   to be labeled with its own Timestamp (<a href="#section-6.36">Section 6.36</a>), returned in the
   response. The server caches the result until it can be sure that the
   client will not retransmit the same request again.



<span class="grey">Handley, et al.             Standards Track                    [Page 90]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-91" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Each server in a proxy chain generates its own final response to a
   CANCEL request. The server responds immediately upon receipt of the
   CANCEL request rather than waiting until it has received final
   responses from the CANCEL requests it generates.

   BYE and OPTIONS final responses are generated by redirect and user
   agent servers; REGISTER final responses are generated by registrars.
   Note that in contrast to the reliability mechanism described in
   <a href="#section-10.5">Section 10.5</a>, responses to these requests are not retransmitted
   periodically and not acknowledged via ACK.

<span class="h4"><a class="selflink" id="section-10.4.2" href="#section-10.4.2">10.4.2</a> TCP</span>

   Clients using TCP do not need to retransmit requests.

<span class="h3"><a class="selflink" id="section-10.5" href="#section-10.5">10.5</a> Reliability for INVITE Requests</span>

   Special considerations apply for the INVITE method.

        1.   After receiving an invitation, considerable time can elapse
             before the server can determine the outcome. For example,
             if the called party is "rung" or extensive searches are
             performed, delays between the request and a definitive
             response can reach several tens of seconds. If either
             caller or callee are automated servers not directly
             controlled by a human being, a call attempt could be
             unbounded in time.

        2.   If a telephony user interface is modeled or if we need to
             interface to the PSTN, the caller's user interface will
             provide "ringback", a signal that the callee is being
             alerted. (The status response 180 (Ringing) MAY be used to
             initiate ringback.) Once the callee picks up, the caller
             needs to know so that it can enable the voice path and stop
             ringback. The callee's response to the invitation could get
             lost. Unless the response is transmitted reliably, the
             caller will continue to hear ringback while the callee
             assumes that the call exists.

        3.   The client has to be able to terminate an on-going request,
             e.g., because it is no longer willing to wait for the
             connection or search to succeed. The server will have to
             wait several retransmission intervals to interpret the lack
             of request retransmissions as the end of a call. If the
             call succeeds shortly after the caller has given up, the
             callee will "pick up the phone" and not be "connected".





<span class="grey">Handley, et al.             Standards Track                    [Page 91]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-92" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-10.5.1" href="#section-10.5.1">10.5.1</a> UDP</span>

   For UDP, A SIP client SHOULD retransmit a SIP INVITE request with an
   interval that starts at T1 seconds, and doubles after each packet
   transmission. The client ceases retransmissions if it receives a
   provisional or definitive response, or once it has sent a total of 7
   request packets.

   A server which transmits a provisional response should retransmit it
   upon reception of a duplicate request. A server which transmits a
   final response should retransmit it with an interval that starts at
   T1 seconds, and doubles for each subsequent packet. Response
   retransmissions cease when any one of the following occurs:

        1.   An ACK request for the same transaction is received;

        2.   a BYE request for the same call leg is received;

        3.   a CANCEL request for the same call leg is received and the
             final response status was equal or greater to 300;

        4.   the response has been transmitted 7 times.

   Only the user agent client generates an ACK for 2xx final responses,
   If the response contained a Contact header field, the ACK MAY be sent
   to the address listed in that Contact header field. If the response
   did not contain a Contact header, the client uses the same To header
   field and Request-URI as for the INVITE request and sends the ACK to
   the same destination as the original INVITE request. ACKs for final
   responses other than 2xx are sent to the same server that the
   original request was sent to, using the same Request-URI as the
   original request. Note, however, that the To header field in the ACK
   is copied from the response being acknowledged, not the request, and
   thus MAY additionally contain the tag parameter. Also note than
   unlike 2xx final responses, a proxy generates an ACK for non-2xx
   final responses.

   The ACK request MUST NOT be acknowledged to prevent a response-ACK
   feedback loop. Fig. 12 and 13 show the client and server state
   diagram for invitations.




        The mechanism in Sec. 10.4 would not work well for INVITE
        because of the long delays between INVITE and a final
        response. If the 200 response were to get lost, the callee
        would believe the call to exist, but the voice path would



<span class="grey">Handley, et al.             Standards Track                    [Page 92]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-93" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




              +===========+
              *           *
  ...........&gt;*  Initial  *&lt;;;;;;;;;;;
  : 7 INVITE  *           *          ;
  :   sent    +===========+          ;
  :                 |                ;
  :                 |    -           ;
  :                 |  INVITE        ;
  :                 |                ;
  :                 v                ;
  :           *************          ;
  : T1*2^n &lt;--*           *          ;
  : INVITE --&gt;*  Calling  *--------+ ;
  :           *           *        | ;
  :           *************        | ;
  :             :   |              | ;
  :.............:   | 1xx      xxx | ;
                    |  -       ACK | ;
                    |              | ;
                    v              | ;
              *************        | ;
              *           *        | ;
              *  Ringing  *&lt;-&gt;1xx  | ;
              *           *        | ;
              *************        | ;
                    |              | ;
                    |&lt;-------------+ ;
                    |                ;
                    v                ;
              *************          ;
      xxx  &lt;--*           *          ;
      ACK  --&gt;* Completed *          ;
              *           *          ;
              *************          ;
                    ; 32s (for proxy);
                    ;;;;;;;;;;;;;;;;;;

 event (xxx=status)
     message


   Figure 12: State transition diagram of client for INVITE method







<span class="grey">Handley, et al.             Standards Track                    [Page 93]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-94" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>




   7 pkts sent  +===============+
+--------------&gt;*               *
|               *   Initial     *&lt;...............
|;;;;;;;;;;;;;;&gt;*               *               :
|;              +===============+               :
|; CANCEL               !                       :
|;  200                 !  INVITE               :
|;                      !   1xx                 :
|;                      !                       :
|;                      v                       :
|;              *****************          BYE  :
|;    INVITE --&gt;*               *          200  :
|;      1xx  &lt;--* Call proceed. *..............&gt;:
|;              *               *               :
|;;;;;;;;;;;;;;;*****************               :
|;                    !   !                     :
|:                    !   !                     :
|;         failure    !   !  picks up           :
|;         &gt;= 300     !   !    200              :
|;            +-------+   +-------+             :
|;            v                   v             :
|;       ***********         ***********        :
|;INVITE&lt;*         *&lt;T1*2^n-&gt;*         *&gt;INVITE :
|;status&gt;* failure *&gt;status&lt;-* success *&lt;status :
|;       *         *         *         *        :
|;;;;;;;;***********         ***********        :
|             ! : |            |  !  :          :
|             ! : |            |  !  :          :
+-------------!-:-+------------+  !  :          :
              ! :.................!..:.........&gt;:
              !                   !         BYE :
              +---------+---------+         200 :
  event                 ! ACK                   :
message sent            v                       :
                *****************               :
            V---*               *               :
           ACK  *   Confirmed   *               :
            |--&gt;*               *               :
                *****************               .
                        :......................&gt;:


   Figure 13: State transition diagram of server for INVITE method

<span class="grey">Handley, et al.             Standards Track                    [Page 94]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-95" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        be dead since the caller does not know that the callee has
        picked up. Thus, the INVITE retransmission interval would
        have to be on the order of a second or two to limit the
        duration of this state confusion. Retransmitting the
        response with an exponential back-off helps ensure that the
        response is received, without placing an undue burden on
        the network.

<span class="h4"><a class="selflink" id="section-10.5.2" href="#section-10.5.2">10.5.2</a> TCP</span>

   A user agent using TCP MUST NOT retransmit requests, but uses the
   same algorithm as for UDP (<a href="#section-10.5.1">Section 10.5.1</a>) to retransmit responses
   until it receives an ACK.


        It is necessary to retransmit 2xx responses as their
        reliability is assured end-to-end only. If the chain of
        proxies has a UDP link in the middle, it could lose the
        response, with no possibility of recovery. For simplicity,
        we also retransmit non-2xx responses, although that is not
        strictly necessary.

<span class="h3"><a class="selflink" id="section-10.6" href="#section-10.6">10.6</a> Reliability for ACK Requests</span>

   The ACK request does not generate responses. It is only generated
   when a response to an INVITE request arrives (see <a href="#section-10.5">Section 10.5</a>). This
   behavior is independent of the transport protocol. Note that the ACK
   request MAY take a different path than the original INVITE request,
   and MAY even cause a new TCP connection to be opened in order to send
   it.

<span class="h3"><a class="selflink" id="section-10.7" href="#section-10.7">10.7</a> ICMP Handling</span>

   Handling of ICMP messages in the case of UDP messages is
   straightforward. For requests, a host, network, port, or protocol
   unreachable error SHOULD be treated as if a 400-class response was
   received. For responses, these errors SHOULD cause the server to
   cease retransmitting the response.

   Source quench ICMP messages SHOULD be ignored. TTL exceeded errors
   SHOULD be ignored. Parameter problem errors SHOULD be treated as if a
   400-class response was received.

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a> Behavior of SIP User Agents</span>

   This section describes the rules for user agent client and servers
   for generating and processing requests and responses.




<span class="grey">Handley, et al.             Standards Track                    [Page 95]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-96" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a> Caller Issues Initial INVITE Request</span>

   When a user agent client desires to initiate a call, it formulates an
   INVITE request. The To field in the request contains the address of
   the callee. The Request-URI contains the same address. The From field
   contains the address of the caller.  If the From address can appear
   in requests generated by other user agent clients for the same call,
   the caller MUST insert the tag parameter in the From field. A UAC MAY
   optionally add a Contact header containing an address where it would
   like to be contacted for transactions from the callee back to the
   caller.

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a> Callee Issues Response</span>

   When the initial INVITE request is received at the callee, the callee
   can accept, redirect, or reject the call. In all of these cases, it
   formulates a response. The response MUST copy the To, From, Call-ID,
   CSeq and Via fields from the request. Additionally, the responding
   UAS MUST add the tag parameter to the To field in the response if the
   request contained more than one Via header field. Since a request
   from a UAC may fork and arrive at multiple hosts, the tag parameter
   serves to distinguish, at the UAC, multiple responses from different
   UAS's. The UAS MAY add a Contact header field in the response. It
   contains an address where the callee would like to be contacted for
   subsequent transactions, including the ACK for the current INVITE.
   The UAS stores the values of the To and From field, including any
   tags. These become the local and remote addresses of the call leg,
   respectively.

<span class="h3"><a class="selflink" id="section-11.3" href="#section-11.3">11.3</a> Caller Receives Response to Initial Request</span>

   Multiple responses may arrive at the UAC for a single INVITE request,
   due to a forking proxy. Each response is distinguished by the "tag"
   parameter in the To header field, and each represents a distinct call
   leg. The caller MAY choose to acknowledge or terminate the call with
   each responding UAS. To acknowledge, it sends an ACK request, and to
   terminate it sends a BYE request.  The To header field in the ACK or
   BYE MUST be the same as the To field in the 200 response, including
   any tag. The From header field MUST be the same as the From header
   field in the 200 (OK) response, including any tag. The Request-URI of
   the ACK or BYE request MAY be set to whatever address was found in
   the Contact header field in the 200 (OK) response, if present.
   Alternately, a UAC may copy the address from the To header field into
   the Request-URI. The UAC also notes the value of the To and From
   header fields in each response. For each call leg, the To header
   field becomes the remote address, and the From header field becomes
   the local address.




<span class="grey">Handley, et al.             Standards Track                    [Page 96]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-97" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-11.4" href="#section-11.4">11.4</a> Caller or Callee Generate Subsequent Requests</span>

   Once the call has been established, either the caller or callee MAY
   generate INVITE or BYE requests to change or terminate the call.
   Regardless of whether the caller or callee is generating the new
   request, the header fields in the request are set as follows. For the
   desired call leg, the To header field is set to the remote address,
   and the From header field is set to the local address (both including
   any tags). The Contact header field MAY be different than the Contact
   header field sent in a previous response or request. The Request-URI
   MAY be set to the value of the Contact header field received in a
   previous request or response from the remote party, or to the value
   of the remote address.

<span class="h3"><a class="selflink" id="section-11.5" href="#section-11.5">11.5</a> Receiving Subsequent Requests</span>

   When a request is received subsequently, the following checks are
   made:

        1.   If the Call-ID is new, the request is for a new call,
             regardless of the values of the To and From header fields.

        2.   If the Call-ID exists, the request is for an existing call.
             If the To, From, Call-ID, and CSeq values exactly match
             (including tags) those of any requests received previously,
             the request is a retransmission.

        3.   If there was no match to the previous step, the To and From
             fields are compared against existing call leg local and
             remote addresses. If there is a match, and the CSeq in the
             request is higher than the last CSeq received on that leg,
             the request is a new transaction for an existing call leg.

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a> Behavior of SIP Proxy and Redirect Servers</span>

   This section describes behavior of SIP redirect and proxy servers in
   detail. Proxy servers can "fork" connections, i.e., a single incoming
   request spawns several outgoing (client) requests.

<span class="h3"><a class="selflink" id="section-12.1" href="#section-12.1">12.1</a> Redirect Server</span>

   A redirect server does not issue any SIP requests of its own. After
   receiving a request other than CANCEL, the server gathers the list of
   alternative locations and returns a final response of class 3xx or it
   refuses the request. For well-formed CANCEL requests, it SHOULD
   return a 2xx response. This response ends the SIP transaction. The





<span class="grey">Handley, et al.             Standards Track                    [Page 97]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-98" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   redirect server maintains transaction state for the whole SIP
   transaction. It is up to the client to detect forwarding loops
   between redirect servers.

<span class="h3"><a class="selflink" id="section-12.2" href="#section-12.2">12.2</a> User Agent Server</span>

   User agent servers behave similarly to redirect servers, except that
   they also accept requests and can return a response of class 2xx.

<span class="h3"><a class="selflink" id="section-12.3" href="#section-12.3">12.3</a> Proxy Server</span>

   This section outlines processing rules for proxy servers. A proxy
   server can either be stateful or stateless. When stateful, a proxy
   remembers the incoming request which generated outgoing requests, and
   the outgoing requests. A stateless proxy forgets all information once
   an outgoing request is generated. A forking proxy SHOULD be stateful.
   Proxies that accept TCP connections MUST be stateful.


        Otherwise, if the proxy were to lose a request, the TCP
        client would never retransmit it.

   A stateful proxy SHOULD NOT become stateless until after it sends a
   definitive response upstream, and at least 32 seconds after it
   received a definitive response.

   A stateful proxy acts as a virtual UAS/UAC. It implements the server
   state machine when receiving requests, and the client state machine
   for generating outgoing requests, with the exception of receiving a
   2xx response to an INVITE. Instead of generating an ACK, the 2xx
   response is always forwarded upstream towards the caller.
   Furthermore, ACK's for 200 responses to INVITE's are always proxied
   downstream towards the UAS, as they would be for a stateless proxy.

   A stateless proxy does not act as a virtual UAS/UAC (as this would
   require state). Rather, a stateless proxy forwards every request it
   receives downstream, and every response it receives upstream.

<span class="h4"><a class="selflink" id="section-12.3.1" href="#section-12.3.1">12.3.1</a> Proxying Requests</span>

   To prevent loops, a server MUST check if its own address is already
   contained in the Via header field of the incoming request.

   The To, From, Call-ID, and Contact tags are copied exactly from the
   original request. The proxy SHOULD change the Request-URI to indicate
   the server where it intends to send the request.





<span class="grey">Handley, et al.             Standards Track                    [Page 98]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-99" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   A proxy server always inserts a Via header field containing its own
   address into those requests that are caused by an incoming request.
   Each proxy MUST insert a "branch" parameter (<a href="#section-6.40">Section 6.40</a>).

<span class="h4"><a class="selflink" id="section-12.3.2" href="#section-12.3.2">12.3.2</a> Proxying Responses</span>

   A proxy only processes a response if the topmost Via field matches
   one of its addresses. A response with a non-matching top Via field
   MUST be dropped.

<span class="h4"><a class="selflink" id="section-12.3.3" href="#section-12.3.3">12.3.3</a> Stateless Proxy: Proxying Responses</span>

   A stateless proxy removes its own Via field, and checks the address
   in the next Via field. In the case of UDP, the response is sent to
   the address listed in the "maddr" tag if present, otherwise to the
   "received" tag if present, and finally to the address in the "sent-
   by" field. A proxy MUST remain stateful when handling requests
   received via TCP.

   A stateless proxy MUST NOT generate its own provisional responses.

<span class="h4"><a class="selflink" id="section-12.3.4" href="#section-12.3.4">12.3.4</a> Stateful Proxy: Receiving Requests</span>

   When a stateful proxy receives a request, it checks the To, From
   (including tags), Call-ID and CSeq against existing request records.
   If the tuple exists, the request is a retransmission. The provisional
   or final response sent previously is retransmitted, as per the server
   state machine. If the tuple does not exist, the request corresponds
   to a new transaction, and the request should be proxied.

   A stateful proxy server MAY generate its own provisional (1xx)
   responses.

<span class="h4"><a class="selflink" id="section-12.3.5" href="#section-12.3.5">12.3.5</a> Stateful Proxy: Receiving ACKs</span>

   When an ACK request is received, it is either processed locally or
   proxied. To make this determination, the To, From, CSeq and Call-ID
   fields are compared against those in previous requests. If there is
   no match, the ACK request is proxied as if it were an INVITE request.
   If there is a match, and if the server had ever sent a 200 response
   upstream, the ACK is proxied.  If the server had never sent any
   responses upstream, the ACK is also proxied. If the server had sent a
   3xx, 4xx, 5xx or 6xx response, but no 2xx response, the ACK is
   processed locally if the tag in the To field of the ACK matches the
   tag sent by the proxy in the response.






<span class="grey">Handley, et al.             Standards Track                    [Page 99]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-100" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-12.3.6" href="#section-12.3.6">12.3.6</a> Stateful Proxy: Receiving Responses</span>

   When a proxy server receives a response that has passed the Via
   checks, the proxy server checks the To (without the tag), From
   (including the tag), Call-ID and CSeq against values seen in previous
   requests. If there is no match, the response is forwarded upstream to
   the address listed in the Via field. If there is a match, the
   "branch" tag in the Via field is examined. If it matches a known
   branch identifier, the response is for the given branch, and
   processed by the virtual client for the given branch. Otherwise, the
   response is dropped.

   A stateful proxy should obey the rules in <a href="#section-12.4">Section 12.4</a> to determine
   if the response should be proxied upstream. If it is to be proxied,
   the same rules for stateless proxies above are followed, with the
   following addition for TCP. If a request was received via TCP
   (indicated by the protocol in the top Via header), the proxy checks
   to see if it has a connection currently open to that address. If so,
   the response is sent on that connection.  Otherwise, a new TCP
   connection is opened to the address and port in the Via field, and
   the response is sent there. Note that this implies that a UAC or
   proxy MUST be prepared to receive responses on the incoming side of a
   TCP connection. Definitive non 200-class responses MUST be
   retransmitted by the proxy, even over a TCP connection.

<span class="h4"><a class="selflink" id="section-12.3.7" href="#section-12.3.7">12.3.7</a> Stateless, Non-Forking Proxy</span>

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   However, servers MAY choose to always operate in a mode that allows
   issuing of several requests, as described in <a href="#section-12.4">Section 12.4</a>.

   The server can forward the request and any responses. It does not
   have to maintain any state for the SIP transaction. Reliability is
   assured by the next redirect or stateful proxy server in the server
   chain.

   A proxy server SHOULD cache the result of any address translations
   and the response to speed forwarding of retransmissions. After the
   cache entry has been expired, the server cannot tell whether an
   incoming request is actually a retransmission of an older request.
   The server will treat it as a new request and commence another
   search.

<span class="h3"><a class="selflink" id="section-12.4" href="#section-12.4">12.4</a> Forking Proxy</span>

   The server MUST respond to the request immediately with a 100
   (Trying) response.



<span class="grey">Handley, et al.             Standards Track                   [Page 100]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-101" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Successful responses to an INVITE request MAY contain a Contact
   header field so that the following ACK or BYE bypasses the proxy
   search mechanism. If the proxy requires future requests to be routed
   through it, it adds a Record-Route header to the request (<a href="#section-6.29">Section</a>
   <a href="#section-6.29">6.29</a>).

   The following C-code describes the behavior of a proxy server issuing
   several requests in response to an incoming INVITE request.  The
   function request(r, a, b) sends a SIP request of type r to address a,
   with branch id b. await_response() waits until a response is received
   and returns the response. close(a) closes the TCP connection to
   client with address a. response(r) sends a response to the client.
   ismulticast() returns 1 if the location is a multicast address and
   zero otherwise.  The variable timeleft indicates the amount of time
   left until the maximum response time has expired. The variable
   recurse indicates whether the server will recursively try addresses
   returned through a 3xx response. A server MAY decide to recursively
   try only certain addresses, e.g., those which are within the same
   domain as the proxy server. Thus, an initial multicast request can
   trigger additional unicast requests.


     /* request type */
     typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;

     process_request(Method R, int N, address_t address[])
     {
       struct {
         int branch;         /* branch id */
         int done;           /* has responded */
       } outgoing[];
       int done[];           /* address has responded */
       char *location[];     /* list of locations */
       int heard = 0;        /* number of sites heard from */
       int class;            /* class of status code */
       int timeleft = 120;   /* sample timeout value */
       int loc = 0;          /* number of locations */
       struct {              /* response */
         int status;         /* response: CANCEL=-1 */
         int locations;      /* number of redirect locations */
         char *location[];   /* redirect locations */
         address_t a;        /* address of respondent */
         int branch;         /* branch identifier */
       } r, best;            /* response, best response */
       int i;

       best.status = 1000;
       for (i = 0; i &lt; N; i++) {



<span class="grey">Handley, et al.             Standards Track                   [Page 101]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-102" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


         request(R, address[i], i);
         outgoing[i].done = 0;
         outgoing[i].branch = i;
       }

       while (timeleft &gt; 0 &amp;&amp; heard &lt; N) {
         r = await_response();
         class = r.status / 100;

         /* If final response, mark branch as done. */
         if (class &gt;= 2) {
           heard++;
           for (i = 0; i &lt; N; i++) {
             if (r.branch == outgoing[i].branch) {
               outgoing[i].done = 1;
               break;
             }
           }
         }
         /* CANCEL: respond, fork and wait for responses */
         else if (class &lt; 0) {
           best.status = 200;
           response(best);
           for (i = 0; i &lt; N; i++) {
             if (!outgoing[i].done)
               request(CANCEL, address[i], outgoing[i].branch);
           }
           best.status = -1;
         }

         /* Send an ACK */

         if (class != 2) {
           if (R == INVITE) request(ACK, r.a, r.branch);
         }


         if (class == 2) {
           if (r.status &lt; best.status) best = r;
           break;
         }
         else if (class == 3) {
           /* A server MAY optionally recurse.  The server MUST check
            * whether it has tried this location before and whether
            * the location is part of the Via path of the incoming
            * request.  This check is omitted here for brevity.
            * Multicast locations MUST NOT be returned to the client if
            * the server is not recursing.



<span class="grey">Handley, et al.             Standards Track                   [Page 102]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-103" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


            */
           if (recurse) {
             multicast = 0;
             N += r.locations;
             for (i = 0; i &lt; r.locations; i++) {
               request(R, r.location[i]);
             }
           } else if (!ismulticast(r.location)) {
             best = r;
           }
         }
         else if (class == 4) {
           if (best.status &gt;= 400) best = r;
         }
         else if (class == 5) {
           if (best.status &gt;= 500) best = r;
         }
         else if (class == 6) {
           best = r;
           break;
         }
       }

       /* We haven't heard anything useful from anybody. */
       if (best.status == 1000) {
         best.status = 404;
       }
       if (best.status/100 != 3) loc = 0;
       response(best);
     }


   Responses are processed as follows. The process completes (and state
   can be freed) when all requests have been answered by final status
   responses (for unicast) or 60 seconds have elapsed (for multicast). A
   proxy MAY send a CANCEL to all branches and return a 408 (Timeout) to
   the client after 60 seconds or more.

   1xx: The proxy MAY forward the response upstream towards the client.

   2xx: The proxy MUST forward the response upstream towards the client,
        without sending an ACK downstream. After receiving a 2xx, the
        server MAY terminate all other pending requests by sending a
        CANCEL request and closing the TCP connection, if applicable.
        (Terminating pending requests is advisable as searches consume
        resources. Also, INVITE requests could "ring" on a number of
        workstations if the callee is currently logged in more than
        once.)



<span class="grey">Handley, et al.             Standards Track                   [Page 103]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-104" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   3xx: The proxy MUST send an ACK and MAY recurse on the listed Contact
        addresses. Otherwise, the lowest-numbered response is returned
        if there were no 2xx responses.

        Location lists are not merged as that would prevent
        forwarding of authenticated responses. Also, responses can
        have message bodies, so that merging is not feasible.

   4xx, 5xx: The proxy MUST send an ACK and remember the response if it
        has a lower status code than any previous 4xx and 5xx responses.
        On completion, the lowest-numbered response is returned if there
        were no 2xx or 3xx responses.

   6xx: The proxy MUST forward the response to the client and send an
        ACK. Other pending requests MAY be terminated with CANCEL as
        described for 2xx responses.

   A proxy server forwards any response for Call-IDs for which it does
   not have a pending transaction according to the response's Via
   header. User agent servers respond to BYE requests for unknown call
   legs with status code 481 (Transaction Does Not Exist); they drop ACK
   requests with unknown call legs silently.

   Special considerations apply for choosing forwarding destinations for
   ACK and BYE requests. In most cases, these requests will bypass
   proxies and reach the desired party directly, keeping proxies from
   having to make forwarding decisions.

   A proxy MAY maintain call state for a period of its choosing. If a
   proxy still has list of destinations that it forwarded the last
   INVITE to, it SHOULD direct ACK requests only to those downstream
   servers.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a> Security Considerations</span>

<span class="h3"><a class="selflink" id="section-13.1" href="#section-13.1">13.1</a> Confidentiality and Privacy: Encryption</span>

<span class="h4"><a class="selflink" id="section-13.1.1" href="#section-13.1.1">13.1.1</a> End-to-End Encryption</span>

   SIP requests and responses can contain sensitive information about
   the communication patterns and communication content of individuals.
   The SIP message body MAY also contain encryption keys for the session
   itself. SIP supports three complementary forms of encryption to
   protect privacy:

        o  End-to-end encryption of the SIP message body and certain
          sensitive header fields;




<span class="grey">Handley, et al.             Standards Track                   [Page 104]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-105" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        o  hop-by-hop encryption to prevent eavesdropping that tracks
          who is calling whom;

        o  hop-by-hop encryption of Via fields to hide the route a
          request has taken.

   Not all of the SIP request or response can be encrypted end-to-end
   because header fields such as To and Via need to be visible to
   proxies so that the SIP request can be routed correctly.  Hop-by-hop
   encryption encrypts the entire SIP request or response on the wire so
   that packet sniffers or other eavesdroppers cannot see who is calling
   whom. Hop-by-hop encryption can also encrypt requests and responses
   that have been end-to-end encrypted. Note that proxies can still see
   who is calling whom, and this information is also deducible by
   performing a network traffic analysis, so this provides a very
   limited but still worthwhile degree of protection.

   SIP Via fields are used to route a response back along the path taken
   by the request and to prevent infinite request loops. However, the
   information given by them can also provide useful information to an
   attacker. <a href="#section-6.22">Section 6.22</a> describes how a sender can request that Via
   fields be encrypted by cooperating proxies without compromising the
   purpose of the Via field.

   End-to-end encryption relies on keys shared by the two user agents
   involved in the request. Typically, the message is sent encrypted
   with the public key of the recipient, so that only that recipient can
   read the message. All implementations SHOULD support PGP-based
   encryption [<a href="#ref-33" title="&quot;PGP message exchange formats&quot;">33</a>] and MAY implement other schemes.

   A SIP request (or response) is end-to-end encrypted by splitting the
   message to be sent into a part to be encrypted and a short header
   that will remain in the clear. Some parts of the SIP message, namely
   the request line, the response line and certain header fields marked
   with "n" in the "enc." column in Table 4 and 5 need to be read and
   returned by proxies and thus MUST NOT be encrypted end-to-end.
   Possibly sensitive information that needs to be made available as
   plaintext include destination address (To) and the forwarding path
   (Via) of the call. The Authorization header field MUST remain in the
   clear if it contains a digital signature as the signature is
   generated after encryption, but MAY be encrypted if it contains
   "basic" or "digest" authentication. The From header field SHOULD
   normally remain in the clear, but MAY be encrypted if required, in
   which case some proxies MAY return a 401 (Unauthorized) status if
   they require a From field.






<span class="grey">Handley, et al.             Standards Track                   [Page 105]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-106" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Other header fields MAY be encrypted or MAY travel in the clear as
   desired by the sender. The Subject, Allow and Content-Type header
   fields will typically be encrypted. The Accept, Accept-Language,
   Date, Expires, Priority, Require, Call-ID, Cseq, and Timestamp header
   fields will remain in the clear.

   All fields that will remain in the clear MUST precede those that will
   be encrypted. The message is encrypted starting with the first
   character of the first header field that will be encrypted and
   continuing through to the end of the message body. If no header
   fields are to be encrypted, encrypting starts with the second CRLF
   pair after the last header field, as shown below. Carriage return and
   line feed characters have been made visible as "$", and the encrypted
   part of the message is outlined.


     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: T. A. Watson &lt;sip:watson@bell-telephone.com&gt;$
     From: A. Bell &lt;sip:a.g.bell@bell-telephone.com&gt;$
     Encryption: PGP version=5.0$
     Content-Length: 224$
     Call-ID: 187602141351@worcester.bell-telephone.com$
     CSeq: 488$
     $
   *******************************************************
   * Subject: Mr. Watson, come here.$                    *
   * Content-Type: application/sdp$                      *
   * $                                                   *
   * v=0$                                                *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$        *
   * c=IN IP4 135.180.144.94$                            *
   * m=audio 3456 RTP/AVP 0 3 4 5$                       *
   *******************************************************



   An Encryption header field MUST be added to indicate the encryption
   mechanism used. A Content-Length field is added that indicates the
   length of the encrypted body. The encrypted body is preceded by a
   blank line as a normal SIP message body would be.

   Upon receipt by the called user agent possessing the correct
   decryption key, the message body as indicated by the Content-Length
   field is decrypted, and the now-decrypted body is appended to the
   clear-text header fields. There is no need for an additional
   Content-Length header field within the encrypted body because the
   length of the actual message body is unambiguous after decryption.



<span class="grey">Handley, et al.             Standards Track                   [Page 106]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-107" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Had no SIP header fields required encryption, the message would have
   been as below. Note that the encrypted body MUST then include a blank
   line (start with CRLF) to disambiguate between any possible SIP
   header fields that might have been present and the SIP message body.


     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: T. A. Watson &lt;sip:watson@bell-telephone.com&gt;$
     From: A. Bell &lt;a.g.bell@bell-telephone.com&gt;$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$
     $
   *************************************************
   * $                                             *
   * v=0$                                          *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$  *
   * c=IN IP4 135.180.144.94$                      *
   * m=audio 3456 RTP/AVP 0 3 4 5$                 *
   *************************************************



<span class="h4"><a class="selflink" id="section-13.1.2" href="#section-13.1.2">13.1.2</a> Privacy of SIP Responses</span>

   SIP requests can be sent securely using end-to-end encryption and
   authentication to a called user agent that sends an insecure
   response.  This is allowed by the SIP security model, but is not a
   good idea.  However, unless the correct behavior is explicit, it
   would not always be possible for the called user agent to infer what
   a reasonable behavior was. Thus when end-to-end encryption is used by
   the request originator, the encryption key to be used for the
   response SHOULD be specified in the request. If this were not done,
   it might be possible for the called user agent to incorrectly infer
   an appropriate key to use in the response. Thus, to prevent key-
   guessing becoming an acceptable strategy, we specify that a called
   user agent receiving a request that does not specify a key to be used
   for the response SHOULD send that response unencrypted.

   Any SIP header fields that were encrypted in a request SHOULD also be
   encrypted in an encrypted response. Contact response fields MAY be
   encrypted if the information they contain is sensitive, or MAY be
   left in the clear to permit proxies more scope for localized
   searches.






<span class="grey">Handley, et al.             Standards Track                   [Page 107]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-108" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-13.1.3" href="#section-13.1.3">13.1.3</a> Encryption by Proxies</span>

   Normally, proxies are not allowed to alter end-to-end header fields
   and message bodies. Proxies MAY, however, encrypt an unsigned request
   or response with the key of the call recipient.


        Proxies need to encrypt a SIP request if the end system
        cannot perform encryption or to enforce organizational
        security policies.

<span class="h4"><a class="selflink" id="section-13.1.4" href="#section-13.1.4">13.1.4</a> Hop-by-Hop Encryption</span>

   SIP requests and responses MAY also be protected by security
   mechanisms at the transport or network layer. No particular mechanism
   is defined or recommended here. Two possibilities are IPSEC [<a href="#ref-34" title="&quot;Security architecture for the internet protocol&quot;">34</a>] or
   TLS [<a href="#ref-35" title="&quot;The TLS protocol version 1.0,&quot;">35</a>]. The use of a particular mechanism will generally need to be
   specified out of band, through manual configuration, for example.

<span class="h4"><a class="selflink" id="section-13.1.5" href="#section-13.1.5">13.1.5</a> Via field encryption</span>

   When Via header fields are to be hidden, a proxy that receives a
   request containing an appropriate "Hide: hop" header field (as
   specified in <a href="#section-6.22">section 6.22</a>) SHOULD encrypt the header field. As only
   the proxy that encrypts the field will decrypt it, the algorithm
   chosen is entirely up to the proxy implementor. Two methods satisfy
   these requirements:

        o  The server keeps a cache of Via header fields and the
          associated To header field, and replaces the Via header field
          with an index into the cache. On the reverse path, take the
          Via header field from the cache rather than the message.

        This is insufficient to prevent message looping, and so an
        additional ID MUST be added so that the proxy can detect loops.
        This SHOULD NOT normally be the address of the proxy as the goal
        is to hide the route, so instead a sufficiently large random
        number SHOULD be used by the proxy and maintained in the cache.

        It is possible for replies to get directed to the wrong
        originator if the cache entry gets reused, so great care needs
        to be taken to ensure this does not happen.

        o  The server MAY use a secret key to encrypt the Via field, a
          timestamp and an appropriate checksum in any such message with
          the same secret key. The checksum is needed to detect whether
          successful decoding has occurred, and the timestamp is




<span class="grey">Handley, et al.             Standards Track                   [Page 108]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-109" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


          required to prevent possible replay attacks and to ensure that
          no two requests from the same previous hop have the same
          encrypted Via field.  This is the preferred solution.

<span class="h3"><a class="selflink" id="section-13.2" href="#section-13.2">13.2</a> Message Integrity and Access Control: Authentication</span>

   Protective measures need to be taken to prevent an active attacker
   from modifying and replaying SIP requests and responses. The same
   cryptographic measures that are used to ensure the authenticity of
   the SIP message also serve to authenticate the originator of the
   message.  However, the "basic" and "digest" authentication mechanism
   offer authentication only, without message integrity.

   Transport-layer or network-layer authentication MAY be used for hop-
   by-hop authentication. SIP also extends the HTTP WWW-Authenticate
   (<a href="#section-6.42">Section 6.42</a>) and Authorization (<a href="#section-6.11">Section 6.11</a>) header field and
   their Proxy counterparts to include cryptographically strong
   signatures. SIP also supports the HTTP "basic" and "digest" schemes
   (see <a href="#section-14">Section 14</a>) and other HTTP authentication schemes to be defined
   that offer a rudimentary mechanism of ascertaining the identity of
   the caller.


        Since SIP requests are often sent to parties with which no
        prior communication relationship has existed, we do not
        specify authentication based on shared secrets.

   SIP requests MAY be authenticated using the Authorization header
   field to include a digital signature of certain header fields, the
   request method and version number and the payload, none of which are
   modified between client and called user agent. The Authorization
   header field is used in requests to authenticate the request
   originator end-to-end to proxies and the called user agent, and in
   responses to authenticate the called user agent or proxies returning
   their own failure codes. If required, hop-by-hop authentication can
   be provided, for example, by the IPSEC Authentication Header.

   SIP does not dictate which digital signature scheme is used for
   authentication, but does define how to provide authentication using
   PGP in <a href="#section-15">Section 15</a>. As indicated above, SIP implementations MAY also
   use "basic" and "digest" authentication and other authentication
   mechanisms defined for HTTP. Note that "basic" authentication has
   severe security limitations. The following does not apply to these
   schemes.

   To cryptographically sign a SIP request, the order of the SIP header
   fields is important. When an Authorization header field is present,
   it indicates that all header fields following the Authorization



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-110" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   header field have been included in the signature.  Therefore, hop-
   by-hop header fields which MUST or SHOULD be modified by proxies MUST
   precede the Authorization header field as they will generally be
   modified or added-to by proxy servers.  Hop-by-hop header fields
   which MAY be modified by a proxy MAY appear before or after the
   Authorization header. When they appear before, they MAY be modified
   by a proxy. When they appear after, they MUST NOT be modified by a
   proxy. To sign a request, a client constructs a message from the
   request method (in upper case) followed, without LWS, by the SIP
   version number, followed, again without LWS, by the request headers
   to be signed and the message body.  The message thus constructed is
   then signed.

   For example, if the SIP request is to be:

   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   Authorization: PGP version=5.0, signature=...
   From: A. Bell &lt;sip:a.g.bell@bell-telephone.com&gt;
   To: T. A. Watson &lt;sip:watson@bell-telephone.com&gt;
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5



   Then the data block that is signed is:

   INVITESIP/2.0From: A. Bell &lt;sip:a.g.bell@bell-telephone.com&gt;
   To: T. A. Watson &lt;sip:watson@bell-telephone.com&gt;
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5






<span class="grey">Handley, et al.             Standards Track                   [Page 110]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-111" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   Clients wishing to authenticate requests MUST construct the portion
   of the message below the Authorization header using a canonical form.
   This allows a proxy to parse the message, take it apart, and
   reconstruct it, without causing an authentication failure due to
   extra white space, for example. Canonical form consists of the
   following rules:

        o  No short form header fields

        o  Header field names are capitalized as shown in this document

        o  No white space between the header name and the colon

        o  A single space after the colon

        o  Line termination with a CRLF

        o  No line folding

        o  No comma separated lists of header values; each must appear
          as a separate header

        o  Only a single SP between tokens, between tokens and quoted
          strings, and between quoted strings; no SP after last token or
          quoted string

        o  No LWS between tokens and separators, except as described
          above for after the colon in header fields

   Note that if a message is encrypted and authenticated using a digital
   signature, when the message is generated encryption is performed
   before the digital signature is generated. On receipt, the digital
   signature is checked before decryption.

   A client MAY require that a server sign its response by including a
   Require: org.ietf.sip.signed-response request header field. The
   client indicates the desired authentication method via the WWW-
   Authenticate header.

   The correct behavior in handling unauthenticated responses to a
   request that requires authenticated responses is described in <a href="#section-13.2.1">section</a>
   <a href="#section-13.2.1">13.2.1</a>.









<span class="grey">Handley, et al.             Standards Track                   [Page 111]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-112" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-13.2.1" href="#section-13.2.1">13.2.1</a> Trusting responses</span>

   There is the possibility that an eavesdropper listens to requests and
   then injects unauthenticated responses that terminate, redirect or
   otherwise interfere with a call. (Even encrypted requests contain
   enough information to fake a response.)

   Clients need to be particularly careful with 3xx redirection
   responses.  Thus a client receiving, for example, a 301 (Moved
   Permanently) which was not authenticated when the public key of the
   called user agent is known to the client, and authentication was
   requested in the request SHOULD be treated as suspicious. The correct
   behavior in such a case would be for the called-user to form a dated
   response containing the Contact field to be used, to sign it, and
   give this signed stub response to the proxy that will provide the
   redirection. Thus the response can be authenticated correctly. A
   client SHOULD NOT automatically redirect such a request to the new
   location without alerting the user to the authentication failure
   before doing so.

   Another problem might be responses such as 6xx failure responses
   which would simply terminate a search, or "4xx" and "5xx" response
   failures.

   If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
   valid, as they will not terminate a search. However, fake 6xx
   responses from a rogue proxy terminate a search incorrectly. 6xx
   responses SHOULD be authenticated if requested by the client, and
   failure to do so SHOULD cause such a client to ignore the 6xx
   response and continue a search.

   With UDP, the same problem with 6xx responses exists, but also an
   active eavesdropper can generate 4xx and 5xx responses that might
   cause a proxy or client to believe a failure occurred when in fact it
   did not. Typically 4xx and 5xx responses will not be signed by the
   called user agent, and so there is no simple way to detect these
   rogue responses. This problem is best prevented by using hop-by-hop
   encryption of the SIP request, which removes any additional problems
   that UDP might have over TCP.

   These attacks are prevented by having the client require response
   authentication and dropping unauthenticated responses. A server user
   agent that cannot perform response authentication responds using the
   normal Require response of 420 (Bad Extension).







<span class="grey">Handley, et al.             Standards Track                   [Page 112]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-113" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-13.3" href="#section-13.3">13.3</a> Callee Privacy</span>

   User location and SIP-initiated calls can violate a callee's privacy.
   An implementation SHOULD be able to restrict, on a per-user basis,
   what kind of location and availability information is given out to
   certain classes of callers.

<span class="h3"><a class="selflink" id="section-13.4" href="#section-13.4">13.4</a> Known Security Problems</span>

   With either TCP or UDP, a denial of service attack exists by a rogue
   proxy sending 6xx responses. Although a client SHOULD choose to
   ignore such responses if it requested authentication, a proxy cannot
   do so. It is obliged to forward the 6xx response back to the client.
   The client can then ignore the response, but if it repeats the
   request it will probably reach the same rogue proxy again, and the
   process will repeat.

<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a> SIP Authentication using HTTP Basic and Digest Schemes</span>

   SIP implementations MAY use HTTP's basic and digest authentication
   mechanisms to provide a rudimentary form of security. This section
   overviews usage of these mechanisms in SIP. The basic operation is
   almost completely identical to that for HTTP [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>]. This section
   outlines this operation, pointing to [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>] for details, and noting the
   differences when used in SIP.

<span class="h3"><a class="selflink" id="section-14.1" href="#section-14.1">14.1</a> Framework</span>

   The framework for SIP authentication parallels that for HTTP [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>]. In
   particular, the BNF for auth-scheme, auth-param, challenge, realm,
   realm-value, and credentials is identical. The 401 response is used
   by user agent servers in SIP to challenge the authorization of a user
   agent client. Additionally, registrars and redirect servers MAY make
   use of 401 responses for authorization, but proxies MUST NOT, and
   instead MAY use the 407 response. The requirements for inclusion of
   the Proxy-Authenticate, Proxy-Authorization, WWW-Authenticate, and
   Authorization in the various messages is identical to [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>].

   Since SIP does not have the concept of a canonical root URL, the
   notion of protections spaces are interpreted differently for SIP. The
   realm is a protection domain for all SIP URIs with the same value for
   the userinfo, host and port part of the SIP Request-URI. For example:


      INVITE sip:alice.wonderland@example.com SIP/2.0
      WWW-Authenticate:  Basic realm="business"





<span class="grey">Handley, et al.             Standards Track                   [Page 113]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-114" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   and


      INVITE sip:aw@example.com SIP/2.0
      WWW-Authenticate: Basic realm="business"



   define different protection realms according to this rule.

   When a UAC resubmits a request with its credentials after receiving a
   401 or 407 response, it MUST increment the CSeq header field as it
   would normally do when sending an updated request.

<span class="h3"><a class="selflink" id="section-14.2" href="#section-14.2">14.2</a> Basic Authentication</span>

   The rules for basic authentication follow those defined in [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>], but
   with the words "origin server" replaced with "user agent server,
   redirect server , or registrar".

   Since SIP URIs are not hierarchical, the paragraph in [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>] that
   states that "all paths at or deeper than the depth of the last
   symbolic element in the path field of the Request-URI also are within
   the protection space specified by the Basic realm value of the
   current challenge" does not apply for SIP. SIP clients MAY
   preemptively send the corresponding Authorization header with
   requests for SIP URIs within the same protection realm (as defined
   above) without receipt of another challenge from the server.

<span class="h3"><a class="selflink" id="section-14.3" href="#section-14.3">14.3</a> Digest Authentication</span>

   The rules for digest authentication follow those defined in [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>],
   with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
   differences:

        1.   The URI included in the challenge has the following BNF:


             URI  =  SIP-URL


        2.   The BNF for digest-uri-value is:


             digest-uri-value  =  Request-URI ; a defined in <a href="#section-4.3">Section</a>
             <a href="#section-4.3">4.3</a>





<span class="grey">Handley, et al.             Standards Track                   [Page 114]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-115" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        3.   The example procedure for choosing a nonce based on Etag
             does not work for SIP.

        4.   The Authentication-Info and Proxy-Authentication-Info
             fields are not used in SIP.

        5.   The text in [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>] regarding cache operation does not apply
             to SIP.

        6.   [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>] requires that a server check that the URI in the
             request line, and the URI included in the Authorization
             header, point to the same resource. In a SIP context, these
             two URI's may actually refer to different users, due to
             forwarding at some proxy. Therefore, in SIP, a server MAY
             check that the request-uri in the Authorization header
             corresponds to a user that the server is willing to accept
             forwarded or direct calls for.

<span class="h3"><a class="selflink" id="section-14.4" href="#section-14.4">14.4</a> Proxy-Authentication</span>

   The use of the Proxy-Authentication and Proxy-Authorization parallel
   that as described in [<a href="#ref-36" title="&quot;HTTP authentication: Basic and digest access authentication,&quot;">36</a>], with one difference. Proxies MUST NOT add
   the Proxy-Authorization header. 407 responses MUST be forwarded
   upstream towards the client following the procedures for any other
   response. It is the client's responsibility to add the Proxy-
   Authorization header containing credentials for the proxy which has
   asked for authentication.


        If a proxy were to resubmit a request with a Proxy-
        Authorization header field, it would need to increment the
        CSeq in the new request. However, this would mean that the
        UAC which submitted the original request would discard a
        response from the UAS, as the CSeq value would be
        different.

   See sections <a href="#section-6.26">6.26</a> and <a href="#section-6.27">6.27</a> for additional information on usage of
   these fields as they apply to SIP.

<span class="h2"><a class="selflink" id="section-15" href="#section-15">15</a> SIP Security Using PGP</span>

<span class="h3"><a class="selflink" id="section-15.1" href="#section-15.1">15.1</a> PGP Authentication Scheme</span>

   The "pgp" authentication scheme is based on the model that the client
   authenticates itself with a request signed with the client's private
   key. The server can then ascertain the origin of the request if it
   has access to the public key, preferably signed by a trusted third
   party.



<span class="grey">Handley, et al.             Standards Track                   [Page 115]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-116" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h4"><a class="selflink" id="section-15.1.1" href="#section-15.1.1">15.1.1</a> The WWW-Authenticate Response Header</span>



        WWW-Authenticate =  "WWW-Authenticate" ":" "pgp" pgp-challenge
        pgp-challenge    =  * (";" pgp-params )
        pgp-params       =  realm | pgp-version | pgp-algorithm | nonce
        realm            =  "realm" "=" realm-value
        realm-value      =  quoted-string
        pgp-version      =  "version" "="
                             &lt;"&gt; digit *( "." digit ) *letter &lt;"&gt;
        pgp-algorithm    =  "algorithm" "=" ( "md5" | "sha1" | token )
        nonce            =  "nonce" "=" nonce-value
        nonce-value      =  quoted-string



   The meanings of the values of the parameters used above are as
   follows:

   realm: A string to be displayed to users so they know which identity
        to use. This string SHOULD contain at least the name of the host
        performing the authentication and MAY additionally indicate the
        collection of users who might have access. An example might be "
        Users with call-out privileges ".

   pgp-algorithm: The value of this parameter indicates the PGP message
        integrity check (MIC) to be used to produce the signature. If
        this not present it is assumed to be "md5". The currently
        defined values are "md5" for the MD5 checksum, and "sha1" for
        the SHA.1 algorithm.

   pgp-version: The version of PGP that the client MUST use. Common
        values are "2.6.2" and "5.0". The default is 5.0.

   nonce: A server-specified data string which should be uniquely
        generated each time a 401 response is made. It is RECOMMENDED
        that this string be base64 or hexadecimal data.  Specifically,
        since the string is passed in the header lines as a quoted
        string, the double-quote character is not allowed. The contents
        of the nonce are implementation dependent. The quality of the
        implementation depends on a good choice. Since the nonce is used
        only to prevent replay attacks and is signed, a time stamp in
        units convenient to the server is sufficient.







<span class="grey">Handley, et al.             Standards Track                   [Page 116]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-117" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Replay attacks within the duration of the call setup are of
        limited interest, so that timestamps with a resolution of a
        few seconds are often should be sufficient. In that case,
        the server does not have to keep a record of the nonces.

   Example:

   WWW-Authenticate: pgp ;version="5.0"
     ;realm="Your Startrek identity, please" ;algorithm=md5
     ;nonce="913082051"



<span class="h4"><a class="selflink" id="section-15.1.2" href="#section-15.1.2">15.1.2</a> The Authorization Request Header</span>

   The client is expected to retry the request, passing an Authorization
   header line, which is defined as follows.



        Authorization  =  "Authorization" ":" "pgp" *( ";" pgp-response )
        pgp-response   =  realm | pgp-version | pgp-signature
                          | signed-by | nonce
        pgp-signature  =  "signature" "=" quoted-string
        signed-by      =  "signed-by" "=" &lt;"&gt; URI &lt;"&gt;


   The client MUST increment the CSeq header before resubmitting the
   request. The signature MUST correspond to the From header of the
   request unless the signed-by parameter is provided.

   pgp-signature: The PGP ASCII-armored signature [<a href="#ref-33" title="&quot;PGP message exchange formats&quot;">33</a>], as it appears
        between the "BEGIN PGP MESSAGE" and "END PGP MESSAGE"
        delimiters, without the version indication. The signature is
        included without any linebreaks.

   The signature is computed across the nonce (if present), request
   method, request version and header fields following the Authorization
   header and the message body, in the same order as they appear in the
   message. The request method and version are prepended to the header
   fields without any white space. The signature is computed across the
   headers as sent, and the terminating CRLF. The CRLF following the
   Authorization header is NOT included in the signature.

   A server MAY be configured not to generate nonces only if replay
   attacks are not a concern.





<span class="grey">Handley, et al.             Standards Track                   [Page 117]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-118" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


        Not generating nonces avoids the additional set of request,
        401 response and possibly ACK messages and reduces delay by
        one round-trip time.


        Using the ASCII-armored version is about 25% less space-
        efficient than including the binary signature, but it is
        significantly easier for the receiver to piece together.
        Versions of the PGP program always include the full
        (compressed) signed text in their output unless ASCII-
        armored mode ( -sta ) is specified.  Typical signatures are
        about 200 bytes long. -- The PGP signature mechanism allows
        the client to simply pass the request to an external PGP
        program. This relies on the requirement that proxy servers
        are not allowed to reorder or change header fields.

   realm: The realm is copied from the corresponding WWW-Authenticate
        header field parameter.

   signed-by: If and only if the request was not signed by the entity
        listed in the From header, the signed-by header indicates the
        name of the signing entity, expressed as a URI.

   Receivers of signed SIP messages SHOULD discard any end-to-end header
   fields above the Authorization header, as they may have been
   maliciously added en route by a proxy.

   Example:

   Authorization: pgp version="5.0"
     ;realm="Your Startrek identity, please"
     ;nonce="913082051"
     ;signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
     VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
     SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
     =aIrx"



<span class="h3"><a class="selflink" id="section-15.2" href="#section-15.2">15.2</a> PGP Encryption Scheme</span>

   The PGP encryption scheme uses the following syntax:



        Encryption    =  "Encryption" ":" "pgp" pgp-eparams
        pgp-eparams   =  1# ( pgp-version | pgp-encoding )
        pgp-encoding  =  "encoding" "=" "ascii" | token



<span class="grey">Handley, et al.             Standards Track                   [Page 118]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-119" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   encoding: Describes the encoding or "armor" used by PGP. The value
        "ascii" refers to the standard PGP ASCII armor, without the
        lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
        without the version identifier. By default, the encrypted part
        is included as binary.

   Example:

   Encryption: pgp version="2.6.2", encoding="ascii"



<span class="h3"><a class="selflink" id="section-15.3" href="#section-15.3">15.3</a> Response-Key Header Field for PGP</span>



        Response-Key  =  "Response-Key" ":" "pgp" pgp-eparams
        pgp-eparams   =  1# ( pgp-version | pgp-encoding | pgp-key)
        pgp-key       =  "key" "=" quoted-string


   If ASCII encoding has been requested via the encoding parameter, the
   key parameter contains the user's public key as extracted from the
   pgp key ring with the "pgp -kxa user ".

   Example:

   Response-Key: pgp version="2.6.2", encoding="ascii",
     key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
     mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
     sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
     bmVAY3MuY29sdW1iaWEuZWR1Pg==
     =+y19"



<span class="h2"><a class="selflink" id="section-16" href="#section-16">16</a> Examples</span>

   In the following examples, we often omit the message body and the
   corresponding Content-Length and Content-Type headers for brevity.

<span class="h3"><a class="selflink" id="section-16.1" href="#section-16.1">16.1</a> Registration</span>

   A user at host saturn.bell-tel.com registers on start-up, via
   multicast, with the local SIP server named bell-tel.com. In the
   example, the user agent on saturn expects to receive SIP requests on
   UDP port 3890.




<span class="grey">Handley, et al.             Standards Track                   [Page 119]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-120" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   C-&gt;S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Contact: &lt;sip:watson@saturn.bell-tel.com:3890;transport=udp&gt;
         Expires: 7200



   The registration expires after two hours. Any future invitations for
   watson@bell-tel.com arriving at sip.bell-tel.com will now be
   redirected to watson@saturn.bell-tel.com, UDP port 3890.

   If Watson wants to be reached elsewhere, say, an on-line service he
   uses while traveling, he updates his reservation after first
   cancelling any existing locations:


   C-&gt;S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 2 REGISTER
         Contact: *
         Expires: 0

   C-&gt;S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 3 REGISTER
         Contact: sip:tawatson@example.com



   Now, the server will forward any request for Watson to the server at
   example.com, using the Request-URI tawatson@example.com. For the
   server at example.com to reach Watson, he will need to send a
   REGISTER there, or inform the server of his current location through
   some other means.

   It is possible to use third-party registration. Here, the secretary
   jon.diligent registers his boss, T. Watson:




<span class="grey">Handley, et al.             Standards Track                   [Page 120]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-121" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   C-&gt;S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP pluto.bell-tel.com
         From: sip:jon.diligent@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 17320@pluto.bell-tel.com
         CSeq: 1 REGISTER
         Contact: sip:tawatson@example.com



   The request could be sent to either the registrar at bell-tel.com or
   the server at example.com. In the latter case, the server at
   example.com would proxy the request to the address indicated in the
   Request-URI. Then, Max-Forwards header could be used to restrict the
   registration to that server.

<span class="h3"><a class="selflink" id="section-16.2" href="#section-16.2">16.2</a> Invitation to a Multicast Conference</span>

   The first example invites schooler@vlsi.cs.caltech.edu to a multicast
   session. All examples use the Session Description Protocol (SDP) (<a href="./rfc2327">RFC</a>
   <a href="./rfc2327">2327</a> [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]) as the session description format.

<span class="h4"><a class="selflink" id="section-16.2.1" href="#section-16.2.1">16.2.1</a> Request</span>


   C-&gt;S: INVITE sip:schooler@cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley &lt;sip:mjh@isi.edu&gt;
         To: Eve Schooler &lt;sip:schooler@caltech.edu&gt;
         Call-ID: 2963313058@north.east.isi.edu
         CSeq: 1 INVITE
         Subject: SIP will be discussed, too
         Content-Type: application/sdp
         Content-Length: 187

         v=0
         o=user1 53655765 2353687637 IN IP4 128.3.4.5
         s=Mbone Audio
         i=Discussion of Mbone Engineering Issues
         e=mbone@somewhere.com
         c=IN IP4 224.2.0.1/127
         t=0 0
         m=audio 3456 RTP/AVP 0






<span class="grey">Handley, et al.             Standards Track                   [Page 121]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-122" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The From request header above states that the request was initiated
   by mjh@isi.edu and addressed to schooler@caltech.edu (From header
   fields). The Via fields list the hosts along the path from invitation
   initiator (the last element of the list) towards the callee. In the
   example above, the message was last multicast to the administratively
   scoped group 239.128.16.254 with a ttl of 16 from the host
   csvax.cs.caltech.edu. The second Via header field indicates that it
   was originally sent from the host north.east.isi.edu. The Request-URI
   indicates that the request is currently being being addressed to
   schooler@cs.caltech.edu, the local address that csvax looked up for
   the callee.

   In this case, the session description is using the Session
   Description Protocol (SDP), as stated in the Content-Type header.

   The header is terminated by an empty line and is followed by a
   message body containing the session description.

<span class="h4"><a class="selflink" id="section-16.2.2" href="#section-16.2.2">16.2.2</a> Response</span>

   The called user agent, directly or indirectly through proxy servers,
   indicates that it is alerting ("ringing") the called party:


   S-&gt;C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley &lt;sip:mjh@isi.edu&gt;
         To: Eve Schooler &lt;sip:schooler@caltech.edu&gt; ;tag=9883472
         Call-ID: 2963313058@north.east.isi.edu
         CSeq: 1 INVITE



   A sample response to the invitation is given below. The first line of
   the response states the SIP version number, that it is a 200 (OK)
   response, which means the request was successful. The Via headers are
   taken from the request, and entries are removed hop by hop as the
   response retraces the path of the request. A new authentication field
   MAY be added by the invited user's agent if required. The Call-ID is
   taken directly from the original request, along with the remaining
   fields of the request message. The original sense of From field is
   preserved (i.e., it is the session initiator).

   In addition, the Contact header gives details of the host where the
   user was located, or alternatively the relevant proxy contact point
   which should be reachable from the caller's host.



<span class="grey">Handley, et al.             Standards Track                   [Page 122]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-123" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   S-&gt;C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley &lt;sip:mjh@isi.edu&gt;
         To: Eve Schooler &lt;sip:schooler@caltech.edu&gt; ;tag=9883472
         Call-ID: 2963313058@north.east.isi.edu
         CSeq: 1 INVITE
         Contact: sip:es@jove.cs.caltech.edu



   The caller confirms the invitation by sending an ACK request to the
   location named in the Contact header:


   C-&gt;S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley &lt;sip:mjh@isi.edu&gt;
         To: Eve Schooler &lt;sip:schooler@caltech.edu&gt; ;tag=9883472
         Call-ID: 2963313058@north.east.isi.edu
         CSeq: 1 ACK



<span class="h3"><a class="selflink" id="section-16.3" href="#section-16.3">16.3</a> Two-party Call</span>

   For two-party Internet phone calls, the response must contain a
   description of where to send the data. In the example below, Bell
   calls Watson. Bell indicates that he can receive RTP audio codings 0
   (PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).


   C-&gt;S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt;
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Subject: Mr. Watson, come here.
         Content-Type: application/sdp
         Content-Length: ...

         v=0
         o=bell 53655765 2353687637 IN IP4 128.3.4.5
         s=Mr. Watson, come here.
         c=IN IP4 kton.bell-tel.com
         m=audio 3456 RTP/AVP 0 3 4 5



<span class="grey">Handley, et al.             Standards Track                   [Page 123]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-124" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   S-&gt;C: SIP/2.0 100 Trying
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S-&gt;C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S-&gt;C: SIP/2.0 182 Queued, 2 callers ahead
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S-&gt;C: SIP/2.0 182 Queued, 1 caller ahead
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S-&gt;C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Contact: sip:watson@boston.bell-tel.com
         Content-Type: application/sdp
         Content-Length: ...

         v=0
         o=watson 4858949 4858949 IN IP4 192.1.2.3
         s=I'm on my way
         c=IN IP4 boston.bell-tel.com
         m=audio 5004 RTP/AVP 0 3




<span class="grey">Handley, et al.             Standards Track                   [Page 124]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-125" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   The example illustrates the use of informational status responses.
   Here, the reception of the call is confirmed immediately (100), then,
   possibly after some database mapping delay, the call rings (180) and
   is then queued, with periodic status updates.

   Watson can only receive PCMU and GSM. Note that Watson's list of
   codecs may or may not be a subset of the one offered by Bell, as each
   party indicates the data types it is willing to receive. Watson will
   send audio data to port 3456 at c.bell-tel.com, Bell will send to
   port 5004 at boston.bell-tel.com.

   By default, the media session is one RTP session. Watson will receive
   RTCP packets on port 5005, while Bell will receive them on port 3457.

   Since the two sides have agreed on the set of media, Bell confirms
   the call without enclosing another session description:


   C-&gt;S: ACK sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 ACK



<span class="h3"><a class="selflink" id="section-16.4" href="#section-16.4">16.4</a> Terminating a Call</span>

   To terminate a call, caller or callee can send a BYE request:


   C-&gt;S: BYE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To: T. A. Watson &lt;sip:watson@bell-tel.com&gt; ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 2 BYE



   If the callee wants to abort the call, it simply reverses the To and
   From fields. Note that it is unlikely that a BYE from the callee will
   traverse the same proxies as the original INVITE.







<span class="grey">Handley, et al.             Standards Track                   [Page 125]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-126" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-16.5" href="#section-16.5">16.5</a> Forking Proxy</span>

   In this example, Bell (a.g.bell@bell-tel.com) (C), currently seated
   at host c.bell-tel.com wants to call Watson (t.watson@ieee.org). At
   the time of the call, Watson is logged in at two workstations,
   t.watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
   registered with the IEEE proxy server (P) called sip.ieee.org. The
   IEEE server also has a registration for the home machine of Watson,
   at watson@h.bell-tel.com (H), as well as a permanent registration at
   watson@acm.org (A). For brevity, the examples omit the session
   description and Via header fields.

   Bell's user agent sends the invitation to the SIP server for the
   ieee.org domain:


   C-&gt;P: INVITE sip:t.watson@ieee.org SIP/2.0
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE



   The SIP server at ieee.org tries the four addresses in parallel.  It
   sends the following message to the home machine:


   P-&gt;H: INVITE sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE



   This request immediately yields a 404 (Not Found) response, since
   Watson is not currently logged in at home:


   H-&gt;P: SIP/2.0 404 Not Found
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=87454273



<span class="grey">Handley, et al.             Standards Track                   [Page 126]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-127" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE



   The proxy ACKs the response so that host H can stop retransmitting
   it:

   P-&gt;H: ACK sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=87454273
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 ACK



   Also, P attempts to reach Watson through the ACM server:

   P-&gt;A: INVITE sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE



   In parallel, the next attempt proceeds, with an INVITE to X and Y:


   P-&gt;X: INVITE sip:t.watson@x.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE

   P-&gt;Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=4
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE




<span class="grey">Handley, et al.             Standards Track                   [Page 127]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-128" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


   As it happens, both Watson at X and a colleague in the other lab at
   host Y hear the phones ringing and pick up. Both X and Y return 200s
   via the proxy to Bell.


   X-&gt;P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=3
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt; ;tag=192137601
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 INVITE
         Contact:  sip:t.watson@x.bell-tel.com

   Y-&gt;P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=4
         Via:      SIP/2.0/UDP c.bell-tel.com
         Contact:  sip:t.watson@y.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt; ;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 INVITE



   Both responses are forwarded to Bell, using the Via information.  At
   this point, the ACM server is still searching its database. P can now
   cancel this attempt:


   P-&gt;A: CANCEL sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 CANCEL



   The ACM server gladly stops its neural-network database search and
   responds with a 200. The 200 will not travel any further, since P is
   the last Via stop.


   A-&gt;P: SIP/2.0 200 OK
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;



<span class="grey">Handley, et al.             Standards Track                   [Page 128]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-129" ></span>
<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 CANCEL



   Bell gets the two 200 responses from X and Y in short order. Bell's
   reaction now depends on his software. He can either send an ACK to
   both if human intelligence is needed to determine who he wants to
   talk to or he can automatically reject one of the two calls. Here, he
   acknowledges both, separately and directly to the final destination:


   C-&gt;X: ACK sip:t.watson@x.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=192137601
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 ACK

   C-&gt;Y: ACK sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 ACK



   After a brief discussion between Bell with X and Y, it becomes clear
   that Watson is at X. (Note that this is not a three-way call; only
   Bell can talk to X and Y, but X and Y cannot talk to each other.)
   Thus, Bell sends a BYE to Y, which is replied to:


   C-&gt;Y: BYE sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE

   Y-&gt;C: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:       T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE




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<span class="grey"><a href="./rfc2543">RFC 2543</a>            SIP: Session Initiation Protocol          March 1999</span>


<span class="h3"><a class="selflink" id="section-16.6" href="#section-16.6">16.6</a> Redirects</span>

   Replies with status codes 301 (Moved Permanently) or 302 (Moved
   Temporarily) specify another location using the Contact field.
   Continuing our earlier example, the server P at ieee.org decides to
   redirect rather than proxy the request:


   P-&gt;C: SIP/2.0 302 Moved temporarily
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell &lt;sip:a.g.bell@bell-tel.com&gt;
         To:      T. Watson &lt;sip:t.watson@ieee.org&gt;;tag=72538263
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE
         Contact: sip:watson@h.bell-tel.com,
                   sip:watson@acm.org, sip:t.watson@x.bell-tel.com,
                   sip:watson@y.bell-tel.com
         CSeq: 1 INVITE



   As another example, assume Alice (A) wants to delegate her calls to
   Bob (B) while she is on vacation until July 29th, 1998. Any calls
   meant for her will reach Bob with Alice's To field, indicating to him
   what role he is to play. Charlie (C) calls Alice (A), whose server
   returns:


   A-&gt;C: SIP/2.0 302 Moved temporarily
         From: Charlie &lt;sip:charlie@caller.com&gt;
         To: Alice &lt;sip:alice@anywhere.com&gt; ;tag=2332462
         Call-ID: 27182@caller.com
         Contact: sip:bob@anywhere.com
         Expires: Wed, 29 Jul 1998 9:00:00 GMT
         CSeq: 1 INVITE



   Charlie then sends the following request to the SIP server of the
   anywhere.com domain. Note that the server at anywhere.com forwards
   the request to Bob based on the Request-URI.


   C-&gt;B: INVITE sip:bob@anywhere.com SIP/2.0
         From: sip:charlie@caller.com
         To: sip:alice@anywhere.com
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE



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   In the third redirection example, we assume that all outgoing
   requests are directed through a local firewall F at caller.com, with
   Charlie again inviting Alice:


   C-&gt;F: INVITE sip:alice@anywhere.com SIP/2.0
         From: sip:charlie@caller.com
         To: Alice &lt;sip:alice@anywhere.com&gt;
         Call-ID: 27182@caller.com
         CSeq: 1 INVITE



   The local firewall at caller.com happens to be overloaded and thus
   redirects the call from Charlie to a secondary server S:


   F-&gt;C: SIP/2.0 302 Moved temporarily
         From: sip:charlie@caller.com
         To: Alice &lt;sip:alice@anywhere.com&gt;
         Call-ID: 27182@caller.com
         CSeq: 1 INVITE
         Contact: &lt;sip:alice@anywhere.com:5080;maddr=spare.caller.com&gt;



   Based on this response, Charlie directs the same invitation to the
   secondary server spare.caller.com at port 5080, but maintains the
   same Request-URI as before:


   C-&gt;S: INVITE sip:alice@anywhere.com SIP/2.0
         From: sip:charlie@caller.com
         To: Alice &lt;sip:alice@anywhere.com&gt;
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE



<span class="h3"><a class="selflink" id="section-16.7" href="#section-16.7">16.7</a> Negotiation</span>

   An example of a 606 (Not Acceptable) response is:


   S-&gt;C: SIP/2.0 606 Not Acceptable
         From: sip:mjh@isi.edu
         To: &lt;sip:schooler@cs.caltech.edu&gt; ;tag=7434264
         Call-ID: 14142@north.east.isi.edu



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         CSeq: 1 INVITE
         Contact: sip:mjh@north.east.isi.edu
         Warning: 370 "Insufficient bandwidth (only have ISDN)",
           305 "Incompatible media format",
           330 "Multicast not available"
         Content-Type: application/sdp
         Content-Length: 50

         v=0
         s=Let's talk
         b=CT:128
         c=IN IP4 north.east.isi.edu
         m=audio 3456 RTP/AVP 5 0 7
         m=video 2232 RTP/AVP 31



   In this example, the original request specified a bandwidth that was
   higher than the access link could support, requested multicast, and
   requested a set of media encodings. The response states that only 128
   kb/s is available and that (only) DVI, PCM or LPC audio could be
   supported in order of preference.

   The response also states that multicast is not available.  In such a
   case, it might be appropriate to set up a transcoding gateway and
   re-invite the user.

<span class="h3"><a class="selflink" id="section-16.8" href="#section-16.8">16.8</a> OPTIONS Request</span>

   A caller Alice can use an OPTIONS request to find out the
   capabilities of a potential callee Bob, without "ringing" the
   designated address. Bob returns a description indicating that he is
   capable of receiving audio encodings PCM Ulaw (payload type 0), 1016
   (payload type 1), GSM (payload type 3), and SX7300/8000 (dynamic
   payload type 99), and video encodings H.261 (payload type 31) and
   H.263 (payload type 34).


   C-&gt;S: OPTIONS sip:bob@example.com SIP/2.0
         From: Alice &lt;sip:alice@anywhere.org&gt;
         To: Bob &lt;sip:bob@example.com&gt;
         Call-ID: 6378@host.anywhere.org
         CSeq: 1 OPTIONS
         Accept: application/sdp

   S-&gt;C: SIP/2.0 200 OK
         From: Alice &lt;sip:alice@anywhere.org&gt;
         To: Bob &lt;sip:bob@example.com&gt; ;tag=376364382



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         Call-ID: 6378@host.anywhere.org
         Content-Length: 81
         Content-Type: application/sdp

         v=0
         m=audio 0 RTP/AVP 0 1 3 99
         m=video 0 RTP/AVP 31 34
         a=rtpmap:99 SX7300/8000











































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A Minimal Implementation

<span class="h3"><a class="selflink" id="appendix-A.1" href="#appendix-A.1">A.1</a> Client</span>

   All clients MUST be able to generate the INVITE and ACK requests.
   Clients MUST generate and parse the Call-ID, Content-Length,
   Content-Type, CSeq, From and To headers. Clients MUST also parse the
   Require header. A minimal implementation MUST understand SDP (<a href="./rfc2327">RFC</a>
   <a href="./rfc2327">2327</a>, [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]). It MUST be able to recognize the status code classes 1
   through 6 and act accordingly.

   The following capability sets build on top of the minimal
   implementation described in the previous paragraph. In general, each
   capability listed below builds on the ones above it:

   Basic: A basic implementation adds support for the BYE method to
        allow the interruption of a pending call attempt. It includes a
        User-Agent header in its requests and indicates its preferred
        language in the Accept-Language header.

   Redirection: To support call forwarding, a client needs to be able to
        understand the Contact header, but only the SIP-URL part, not
        the parameters.

   Firewall-friendly: A firewall-friendly client understands the Route
        and Record-Route header fields and can be configured to use a
        local proxy for all outgoing requests.

   Negotiation: A client MUST be able to request the OPTIONS method and
        understand the 380 (Alternative Service) status and the Contact
        parameters to participate in terminal and media negotiation. It
        SHOULD be able to parse the Warning response header to provide
        useful feedback to the caller.

   Authentication: If a client wishes to invite callees that require
        caller authentication, it MUST be able to recognize the 401
        (Unauthorized) status code, MUST be able to generate the
        Authorization request header and MUST understand the WWW-
        Authenticate response header.

   If a client wishes to use proxies that require caller authentication,
   it MUST be able to recognize the 407 (Proxy Authentication Required)
   status code, MUST be able to generate the Proxy-Authorization request
   header and understand the Proxy-Authenticate response header.







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<span class="h3"><a class="selflink" id="appendix-A.2" href="#appendix-A.2">A.2</a> Server</span>

   A minimally compliant server implementation MUST understand the
   INVITE, ACK, OPTIONS and BYE requests. A proxy server MUST also
   understand CANCEL. It MUST parse and generate, as appropriate, the
   Call-ID, Content-Length, Content-Type, CSeq, Expires, From, Max-
   Forwards, Require, To and Via headers. It MUST echo the CSeq and
   Timestamp headers in the response. It SHOULD include the Server
   header in its responses.

<span class="h3"><a class="selflink" id="appendix-A.3" href="#appendix-A.3">A.3</a> Header Processing</span>

   Table 6 lists the headers that different implementations support. UAC
   refers to a user-agent client (calling user agent), UAS to a user-
   agent server (called user-agent).

   The fields in the table have the following meaning. Type is as in
   Table 4 and 5. "-" indicates the field is not meaningful to this
   system (although it might be generated by it). "m" indicates the
   field MUST be understood. "b" indicates the field SHOULD be
   understood by a Basic implementation.  "r" indicates the field SHOULD
   be understood if the system claims to understand redirection. "a"
   indicates the field SHOULD be understood if the system claims to
   support authentication. "e" indicates the field SHOULD be understood
   if the system claims to support encryption. "o" indicates support of
   the field is purely optional. Headers whose support is optional for
   all implementations are not shown.
























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                        type  UAC  proxy  UAS  registrar
   _____________________________________________________
   Accept                R     -     o     m      m
   Accept-Encoding       R     -     -     m      m
   Accept-Language       R     -     b     b      b
   Allow                405    o     -     -      -
   Authorization         R     a     o     a      a
   Call-ID               g     m     m     m      m
   Content-Encoding      g     m     -     m      m
   Content-Length        g     m     m     m      m
   Content-Type          g     m     -     m      m
   CSeq                  g     m     m     m      m
   Encryption            g     e     -     e      e
   Expires               g     -     o     o      m
   From                  g     m     o     m      m
   Hide                  R     -     m     -      -
   Contact               R     -     -     -      m
   Contact               r     r     r     -      -
   Max-Forwards          R     -     b     -      -
   Proxy-Authenticate   407    a     -     -      -
   Proxy-Authorization   R     -     a     -      -
   Proxy-Require         R     -     m     -      -
   Require               R     m     -     m      m
   Response-Key          R     -     -     e      e
   Route                 R     -     m     -      -
   Timestamp             g     o     o     m      m
   To                    g     m     m     m      m
   Unsupported           r     b     b     -      -
   User-Agent            g     b     -     b      -
   Via                   g     m     m     m      m
   WWW-Authenticate     401    a     -     -      -


   Table 6: Header Field Processing Requirements

B Usage of the Session Description Protocol (SDP)

   This section describes the use of the Session Description Protocol
   (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-6" title="&quot;SDP: session description protocol&quot;">6</a>]).

<span class="h3"><a class="selflink" id="appendix-B.1" href="#appendix-B.1">B.1</a> Configuring Media Streams</span>

   The caller and callee align their media descriptions so that the nth
   media stream ("m=" line) in the caller's session description
   corresponds to the nth media stream in the callee's description.




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   All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
   payload types to encodings.

        This allows easier migration away from static payload
        types.

   If the callee wants to neither send nor receive a stream offered by
   the caller, the callee sets the port number of that stream to zero in
   its media description.


        There currently is no other way than port zero for the
        callee to refuse a bidirectional stream offered by the
        caller. Both caller and callee need to be aware what media
        tools are to be started.

   For example, assume that the caller Alice has included the following
   description in her INVITE request. It includes an audio stream and
   two bidirectional video streams, using H.261 (payload type 31) and
   MPEG (payload type 32).


   v=0
   o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
   c=IN IP4 host.anywhere.com
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   m=video 51372 RTP/AVP 31
   a=rtpmap:31 H261/90000
   m=video 53000 RTP/AVP 32
   a=rtpmap:32 MPV/90000



   The callee, Bob, does not want to receive or send the first video
   stream, so it returns the media description below:

   v=0
   o=bob 2890844730 2890844730 IN IP4 host.example.com
   c=IN IP4 host.example.com
   m=audio 47920 RTP/AVP 0 1
   a=rtpmap:0 PCMU/8000
   a=rtpmap:1 1016/8000
   m=video 0 RTP/AVP 31
   m=video 53000 RTP/AVP 32
   a=rtpmap:32 MPV/90000





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<span class="h3"><a class="selflink" id="appendix-B.2" href="#appendix-B.2">B.2</a> Setting SDP Values for Unicast</span>

   If a session description from a caller contains a media stream which
   is listed as send (receive) only, it means that the caller is only
   willing to send (receive) this stream, not receive (send). The same
   is true for the callee.

   For receive-only and send-or-receive streams, the port number and
   address in the session description indicate where the media stream
   should be sent to by the recipient of the session description, either
   caller or callee. For send-only streams, the address and port number
   have no significance and SHOULD be set to zero.

   The list of payload types for each media stream conveys two pieces of
   information, namely the set of codecs that the caller or callee is
   capable of sending or receiving, and the RTP payload type numbers
   used to identify those codecs. For receive-only or send-and-receive
   media streams, a caller SHOULD list all of the codecs it is capable
   of supporting in the session description in an INVITE or ACK. For
   send-only streams, the caller SHOULD indicate only those it wishes to
   send for this session. For receive-only streams, the payload type
   numbers indicate the value of the payload type field in RTP packets
   the caller is expecting to receive for that codec type. For send-only
   streams, the payload type numbers indicate the value of the payload
   type field in RTP packets the caller is planning to send for that
   codec type.  For send-and-receive streams, the payload type numbers
   indicate the value of the payload type field the caller expects to
   both send and receive.

   If a media stream is listed as receive-only by the caller, the callee
   lists, in the response, those codecs it intends to use from among the
   ones listed in the request. If a media stream is listed as send-only
   by the caller, the callee lists, in the response, those codecs it is
   willing to receive among the ones listed in the the request. If the
   media stream is listed as both send and receive, the callee lists
   those codecs it is capable of sending or receiving among the ones
   listed by the caller in the INVITE. The actual payload type numbers
   in the callee's session description corresponding to a particular
   codec MUST be the same as the caller's session description.

   If caller and callee have no media formats in common for a particular
   stream, the callee MUST return a session description containing the
   particular "m=" line, but with the port number set to zero, and no
   payload types listed.

   If there are no media formats in common for all streams, the callee
   SHOULD return a 400 response, with a 304 Warning header field.




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<span class="h3"><a class="selflink" id="appendix-B.3" href="#appendix-B.3">B.3</a> Multicast Operation</span>

   The interpretation of send-only and receive-only for multicast media
   sessions differs from that for unicast sessions. For multicast,
   send-only means that the recipient of the session description (caller
   or callee) SHOULD only send media streams to the address and port
   indicated. Receive-only means that the recipient of the session
   description SHOULD only receive media on the address and port
   indicated.

   For multicast, receive and send multicast addresses are the same and
   all parties use the same port numbers to receive media data. If the
   session description provided by the caller is acceptable to the
   callee, the callee can choose not to include a session description or
   MAY echo the description in the response.

   A callee MAY, in the response, return a session description with some
   of the payload types removed, or port numbers set to zero (but no
   other value). This indicates to the caller that the callee does not
   support the given stream or media types which were removed. A callee
   MUST NOT change whether a given stream is send-only, receive-only, or
   send-and-receive.

   If a callee does not support multicast at all, it SHOULD return a 400
   status response and include a 330 Warning.

<span class="h3"><a class="selflink" id="appendix-B.4" href="#appendix-B.4">B.4</a> Delayed Media Streams</span>

   In some cases, a caller may not know the set of media formats which
   it can support at the time it would like to issue an invitation. This
   is the case when the caller is actually a gateway to another protocol
   which performs media format negotiation after call setup. When this
   occurs, a caller MAY issue an INVITE with a session description that
   contains no media lines. The callee SHOULD interpret this to mean
   that the caller wishes to participate in a multimedia session
   described by the session description, but that the media streams are
   not yet known. The callee SHOULD return a session description
   indicating the streams and media formats it is willing to support,
   however. The caller MAY update the session description either in the
   ACK request or in a re-INVITE at a later time, once the streams are
   known.

<span class="h3"><a class="selflink" id="appendix-B.5" href="#appendix-B.5">B.5</a> Putting Media Streams on Hold</span>

   If a party in a call wants to put the other party "on hold", i.e.,
   request that it temporarily stops sending one or more media streams,
   a party re-invites the other by sending an INVITE request with a
   modified session description. The session description is the same as



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   in the original invitation (or response), but the "c" destination
   addresses for the media streams to be put on hold are set to zero
   (0.0.0.0).

<span class="h3"><a class="selflink" id="appendix-B.6" href="#appendix-B.6">B.6</a> Subject and SDP "s=" Line</span>

   The SDP "s=" line and the SIP Subject header field have different
   meanings when inviting to a multicast session. The session
   description line describes the subject of the multicast session,
   while the SIP Subject header field describes the reason for the
   invitation. The example in <a href="#section-16.2">Section 16.2</a> illustrates this point. For
   invitations to two-party sessions, the SDP "s=" line MAY be left
   empty.

<span class="h3"><a class="selflink" id="appendix-B.7" href="#appendix-B.7">B.7</a> The SDP "o=" Line</span>

   The "o=" line is not strictly necessary for two-party sessions, but
   MUST be present to allow re-use of SDP-based tools.

































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C Summary of Augmented BNF

   All of the mechanisms specified in this document are described in
   both prose and an augmented Backus-Naur Form (BNF) similar to that
   used by <a href="./rfc822">RFC 822</a> [<a href="#ref-9" title="&quot;Media stream packetization and synchronization on non-guaranteed quality of service LANs,&quot;">9</a>]. Implementors will need to be familiar with the
   notation in order to understand this specification. The augmented BNF
   includes the following constructs:



        name  =  definition


   The name of a rule is simply the name itself (without any enclosing
   "&lt;" and "&gt;") and is separated from its definition by the equal "="
   character. White space is only significant in that indentation of
   continuation lines is used to indicate a rule definition that spans
   more than one line. Certain basic rules are in uppercase, such as SP,
   LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
   definitions whenever their presence will facilitate discerning the
   use of rule names.


   "literal"


   Quotation marks surround literal text. Unless stated otherwise, the
   text is case-insensitive.


   rule1 | rule2


   Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
   will accept yes or no.


   (rule1 rule2)


   Elements enclosed in parentheses are treated as a single element.
   Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
   elem" and "elem bar elem".








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   *rule


   The character "*" preceding an element indicates repetition. The full
   form is "&lt;n&gt;*&lt;m&gt;element" indicating at least &lt;n&gt; and at most &lt;m&gt;
   occurrences of element. Default values are 0 and infinity so that
   "*(element)" allows any number, including zero; "1*element" requires
   at least one; and "1*2element" allows one or two.


   [<a id="ref-rule">rule</a>]


   Square brackets enclose optional elements; "[foo bar]" is equivalent
   to "*1(foo bar)".


   N rule


   Specific repetition: "&lt;n&gt;(element)" is equivalent to
   "&lt;n&gt;*&lt;n&gt;(element)"; that is, exactly &lt;n&gt; occurrences of (element).
   Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
   alphabetic characters.


   #rule


   A construct "#" is defined, similar to "*", for defining lists of
   elements. The full form is "&lt;n&gt;#&lt;m&gt; element" indicating at least &lt;n&gt;
   and at most &lt;m&gt; elements, each separated by one or more commas (",")
   and OPTIONAL linear white space (LWS). This makes the usual form of
   lists very easy; a rule such as



           ( *LWS element *( *LWS "," *LWS element ))


   can be shown as 1# element. Wherever this construct is used, null
   elements are allowed, but do not contribute to the count of elements
   present. That is, "(element), , (element)" is permitted, but counts
   as only two elements. Therefore, where at least one element is
   required, at least one non-null element MUST be present. Default
   values are 0 and infinity so that "#element" allows any number,
   including zero; "1#element" requires at least one; and "1#2element"
   allows one or two.



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   ; comment


   A semi-colon, set off some distance to the right of rule text, starts
   a comment that continues to the end of line. This is a simple way of
   including useful notes in parallel with the specifications.


   implied *LWS


   The grammar described by this specification is word-based. Except
   where noted otherwise, linear white space (LWS) can be included
   between any two adjacent words (token or quoted-string), and between
   adjacent tokens and separators, without changing the interpretation
   of a field. At least one delimiter (LWS and/or separators) MUST exist
   between any two tokens (for the definition of "token" below), since
   they would otherwise be interpreted as a single token.

<span class="h3"><a class="selflink" id="appendix-C.1" href="#appendix-C.1">C.1</a> Basic Rules</span>

   The following rules are used throughout this specification to
   describe basic parsing constructs. The US-ASCII coded character set
   is defined by ANSI X3.4-1986.


        OCTET     =  &lt;any 8-bit sequence of data&gt;
        CHAR      =  &lt;any US-ASCII character (octets 0 - 127)&gt;
        upalpha   =  "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
                     "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
                     "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
        lowalpha  =  "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
                     "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
                     "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
        alpha     =  lowalpha | upalpha
        digit     =  "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
                     "8" | "9"
        alphanum  =  alpha | digit
        CTL       =  &lt;any US-ASCII control character
                     (octets 0 -- 31) and DEL (127)&gt;
        CR        =  %d13 ; US-ASCII CR, carriage return character
        LF        =  %d10 ; US-ASCII LF, line feed character
        SP        =  %d32 ; US-ASCII SP, space character
        HT        =  %d09 ; US-ASCII HT, horizontal tab character
        CRLF      =  CR LF ; typically the end of a line


   The following are defined in <a href="./rfc2396">RFC 2396</a> [<a href="#ref-12" title="&quot;Uniform resource identifiers (URI): generic syntax&quot;">12</a>] for the SIP URI:



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        unreserved  =  alphanum | mark
        mark        =  "-" | "_" | "." | "!" | "~" | "*" | "'"
                   |   "(" | ")"
        escaped     =  "%" hex hex


   SIP header field values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab. All linear
   white space, including folding, has the same semantics as SP. A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.



        LWS  =  [CRLF] 1*( SP | HT ) ; linear whitespace


   The TEXT-UTF8 rule is only used for descriptive field contents and
   values that are not intended to be interpreted by the message parser.
   Words of *TEXT-UTF8 contain characters from the UTF-8 character set
   (<a href="./rfc2279">RFC 2279</a> [<a href="#ref-21" title="&quot;UTF-8, a transformation format of ISO 10646&quot;">21</a>]). In this regard, SIP differs from HTTP, which uses
   the ISO 8859-1 character set.



        TEXT-UTF8  =  &lt;any UTF-8 character encoding, except CTLs,
                      but including LWS&gt;


   A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
   header field continuation. It is expected that the folding LWS will
   be replaced with a single SP before interpretation of the TEXT-UTF8
   value.

   Hexadecimal numeric characters are used in several protocol elements.



        hex  =  "A" | "B" | "C" | "D" | "E" | "F"
                | "a" | "b" | "c" | "d" | "e" | "f" | digit


   Many SIP header field values consist of words separated by LWS or
   special characters. These special characters MUST be in a quoted
   string to be used within a parameter value.






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        token       =  1*&lt; any CHAR  except CTL's  or separators&gt;
        separators  =  "(" | ")" | "&lt;" | "&gt;" | "@" |
                       "," | ";" | ":" | "\" | &lt;"&gt; |
                       "/" | "[" | "]" | "?" | "=" |
                       "{" | "}" | SP | HT


   Comments can be included in some SIP header fields by surrounding the
   comment text with parentheses. Comments are only allowed in fields
   containing "comment" as part of their field value definition. In all
   other fields, parentheses are considered part of the field value.



        comment  =  "(" *(ctext | quoted-pair | comment) ")"
        ctext    =  &lt; any TEXT-UTF8  excluding "("  and ")"&gt;


   A string of text is parsed as a single word if it is quoted using
   double-quote marks.



        quoted-string  =  ( &lt;"&gt; *(qdtext | quoted-pair ) &lt;"&gt; )
        qdtext         =  &lt;any TEXT-UTF8 except &lt;"&gt;&gt;


   The backslash character ("\") MAY be used as a single-character
   quoting mechanism only within quoted-string and comment constructs.



        quoted-pair  =  " \ " CHAR


















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D Using SRV DNS Records

   The following procedure is experimental and relies on DNS SRV records
   (<a href="./rfc2052">RFC 2052</a> [<a href="#ref-14" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">14</a>]). The steps listed below are used in place of the two
   steps in <a href="#section-1.4.2">section 1.4.2</a>.

   If a step elicits no addresses, the client continues to the next
   step.  However if a step elicits one or more addresses, but no SIP
   server at any of those addresses responds, then the client concludes
   the server is down and doesn't continue on to the next step.

   When SRV records are to be used, the protocol to use when querying
   for the SRV record is "sip". SRV records contain port numbers for
   servers, in addition to IP addresses; the client always uses this
   port number when contacting the SIP server. Otherwise, the port
   number in the SIP URI is used, if present. If there is no port number
   in the URI, the default port, 5060, is used.

        1.   If the host portion of the Request-URI is an IP address,
             the client contacts the server at the given address. If the
             host portion of the Request-URI is not an IP address, the
             client proceeds to the next step.

        2.   The Request-URI is examined. If it contains an explicit
             port number, the next two steps are skipped.

        3.   The Request-URI is examined. If it does not specify a
             protocol (TCP or UDP), the client queries the name server
             for SRV records for both UDP (if supported by the client)
             and TCP (if supported by the client) SIP servers. The
             format of these queries is defined in <a href="./rfc2052">RFC 2052</a> [<a href="#ref-14" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">14</a>]. The
             results of the query or queries are merged together and
             ordered based on priority. Then, the searching technique
             outlined in <a href="./rfc2052">RFC 2052</a> [<a href="#ref-14" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">14</a>] is used to select servers in
             order.  If DNS doesn't return any records, the user goes to
             the last step.  Otherwise, the user attempts to contact
             each server in the order listed.  If no server is
             contacted, the user gives up.

        4.   If the Request-URI specifies a protocol (TCP or UDP) that
             is supported by the client, the client queries the name
             server for SRV records for SIP servers of that protocol
             type only. If the client does not support the protocol
             specified in the Request-URI, it gives up. The searching
             technique outlined in <a href="./rfc2052">RFC 2052</a> [<a href="#ref-14" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">14</a>] is used to select
             servers from the DNS response in order. If DNS doesn't





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             return any records, the user goes to the last step.
             Otherwise, the user attempts to contact each server in the
             order listed. If no server is contacted, the user gives up.

        5.   The client queries the name server for address records for
             the host portion of the Request-URI. If there were no
             address records, the client stops, as it has been unable to
             locate a server. By address record, we mean A RR's, AAAA
             RR's, or their most modern equivalent.

   A client MAY cache a successful DNS query result. A successful query
   is one which contained records in the answer, and a server was
   contacted at one of the addresses from the answer. When the client
   wishes to send a request to the same host, it starts the search as if
   it had just received this answer from the name server. The server
   uses the procedures specified in <a href="./rfc1035">RFC1035</a> [<a href="#ref-15" title="&quot;Domain names - implementation and specification&quot;">15</a>] regarding cache
   invalidation when the time-to-live of the DNS result expires. If the
   client does not find a SIP server among the addresses listed in the
   cached answer, it starts the search at the beginning of the sequence
   described above.

   For example, consider a client that wishes to send a SIP request. The
   Request-URI for the destination is sip:user@company.com.  The client
   only supports UDP. It would follow these steps:

        1.   The host portion is not an IP address, so the client goes
             to step 2 above.

        2.   The client does a DNS query of QNAME="sip.udp.company.com",
             QCLASS=IN, QTYPE=SRV. Since it doesn't support TCP, it
             omits the TCP query. There were no addresses in the DNS
             response, so the client goes to the next step.

        3.   The client does a DNS query for A records for
             "company.com". An address is found, so that client attempts
             to contact a server at that address at port 5060.















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E IANA Considerations

   <a href="#section-4.4">Section 4.4</a> describes a name space and mechanism for registering SIP
   options.

   <a href="#section-6.41">Section 6.41</a> describes the name space for registering SIP warn-codes.













































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F Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. Detailed comments were provided by Anders
   Kristensen, Jim Buller, Dave Devanathan, Yaron Goland, Christian
   Huitema, Gadi Karmi, Jonathan Lennox, Keith Moore, Vern Paxson, Moshe
   J. Sambol, and Eric Tremblay.

   This work is based, inter alia, on [<a href="#ref-37" title="&quot;Case study: multimedia conference control in a packet-switched teleconferencing system,&quot;">37</a>,<a href="#ref-38" title="&quot;Personal mobility for multimedia services in the Internet,&quot;">38</a>].

G Authors' Addresses

   Mark Handley
   AT&amp;T Center for Internet Research at ISCI (ACIRI)
   1947 Center St., Suite 600
   Berkeley, CA 94704-119
   USA
   Email: mjh@aciri.org

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   Email:  schulzrinne@cs.columbia.edu

   Eve Schooler
   Computer Science Department 256-80
   California Institute of Technology
   Pasadena, CA 91125
   USA
   Email:  schooler@cs.caltech.edu

   Jonathan Rosenberg
   Lucent Technologies, Bell Laboratories
   Rm. 4C-526
   101 Crawfords Corner Road
   Holmdel, NJ 07733
   USA
   Email:  jdrosen@bell-labs.com










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H Bibliography

   [<a id="ref-1">1</a>] Pandya, R., "Emerging mobile and personal communication systems,"
       IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.

   [<a id="ref-2">2</a>] Braden, B., Zhang, L., Berson, S., Herzog, S. and S. Jamin,
       "Resource ReSerVation protocol (RSVP) -- version 1 functional
       specification", <a href="./rfc2205">RFC 2205</a>, October 1997.

   [<a id="ref-3">3</a>] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
       a transport protocol for real-time applications", <a href="./rfc1889">RFC 1889</a>,
       Internet Engineering Task Force, Jan. 1996.

   [<a id="ref-4">4</a>] Schulzrinne, H., Lanphier, R. and A. Rao, "Real time streaming
       protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.

   [<a id="ref-5">5</a>] Handley, M., "SAP: Session announcement protocol," Internet
       Draft, Internet Engineering Task Force, Nov. 1996.  Work in
       progress.

   [<a id="ref-6">6</a>] Handley, M. and V. Jacobson, "SDP: session description protocol",
       <a href="./rfc2327">RFC 2327</a>, April 1998.

   [<a id="ref-7">7</a>] International Telecommunication Union, "Visual telephone systems
       and equipment for local area networks which provide a non-
       guaranteed quality of service," Recommendation H.323,
       Telecommunication Standardization Sector of ITU, Geneva,
       Switzerland, May 1996.

   [<a id="ref-8">8</a>] International Telecommunication Union, "Control protocol for
       multimedia communication," Recommendation H.245,
       Telecommunication Standardization Sector of ITU, Geneva,
       Switzerland, Feb. 1998.

   [<a id="ref-9">9</a>] International Telecommunication Union, "Media stream
       packetization and synchronization on non-guaranteed quality of
       service LANs," Recommendation H.225.0, Telecommunication
       Standardization Sector of ITU, Geneva, Switzerland, Nov. 1996.

   [<a id="ref-10">10</a>] Bradner, S., "Key words for use in RFCs to indicate requirement
        levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>,  <a href="./rfc2119">RFC 2119</a>, Mardch 1997.

   [<a id="ref-11">11</a>] Fielding, R., Gettys, J., Mogul, J., Nielsen, H. and T.
        Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1", <a href="./rfc2068">RFC</a>
        <a href="./rfc2068">2068</a>, January 1997.

   [<a id="ref-12">12</a>] Berners-Lee, T., Fielding, R. and L. Masinter, "Uniform resource
        identifiers (URI): generic syntax", <a href="./rfc2396">RFC 2396</a>, August 1998.



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   [<a id="ref-13">13</a>] Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform resource
        locators (URL)", <a href="./rfc1738">RFC 1738</a>, December 1994.

   [<a id="ref-14">14</a>] Gulbrandsen, A.  and P. Vixie, "A DNS RR for specifying the
        location of services (DNS SRV)", <a href="./rfc2052">RFC 2052</a>, October 1996.

   [<a id="ref-15">15</a>] Mockapetris, P., "Domain names - implementation and
        specification", STD 13, <a href="./rfc1035">RFC 1035</a>, Noveberm 1997.

   [<a id="ref-16">16</a>] Hamilton, M. and R. Wright, "Use of DNS aliases for network
        services", <a href="./rfc2219">RFC 2219</a>, October 1997.

   [<a id="ref-17">17</a>] Zimmerman, D., "The finger user information protocol", <a href="./rfc1288">RFC 1288</a>,
        December 1991.

   [<a id="ref-18">18</a>] Williamson, S., Kosters, M., Blacka, D., Singh, J. and K.
        Zeilstra, "Referral whois (rwhois) protocol V1.5", <a href="./rfc2167">RFC 2167</a>,
        June 1997.

   [<a id="ref-19">19</a>] Yeong, W., Howes, T. and S. Kille, "Lightweight directory access
        protocol", <a href="./rfc1777">RFC 1777</a>, March 1995.

   [<a id="ref-20">20</a>] Schooler, E., "A multicast user directory service for
        synchronous rendezvous," Master's Thesis CS-TR-96-18, Department
        of Computer Science, California Institute of Technology,
        Pasadena, California, Aug. 1996.

   [<a id="ref-21">21</a>] Yergeau, F., "UTF-8, a transformation format of ISO 10646", <a href="./rfc2279">RFC</a>
        <a href="./rfc2279">2279</a>, January 1998.

   [<a id="ref-22">22</a>] Stevens, W., TCP/IP illustrated: the protocols , vol. 1.
        Reading, Massachusetts: Addison-Wesley, 1994.

   [<a id="ref-23">23</a>] Mogul, J. and S. Deering, "Path MTU discovery", <a href="./rfc1191">RFC 1191</a>,
        November 1990.

   [<a id="ref-24">24</a>] Crocker, D., "Standard for the format of ARPA internet text
        messages", RFC STD 11, <a href="./rfc822">RFC 822</a>, August 1982.

   [<a id="ref-25">25</a>] Meyer, D., "Administratively scoped IP multicast", <a href="./rfc2365">RFC 2365</a>,
        July 1998.

   [<a id="ref-26">26</a>] Schulzrinne, H., "RTP profile for audio and video conferences
        with minimal control", <a href="./rfc1890">RFC 1890</a>, January 1996

   [<a id="ref-27">27</a>] Eastlake, D., Crocker, S. and J. Schiller, "Randomness
        recommendations for security", <a href="./rfc1750">RFC 1750</a>, December 1994.




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   [<a id="ref-28">28</a>] Hoffman, P., Masinter, L. and J. Zawinski, "The mailto URL
        scheme", <a href="./rfc2368">RFC 2368</a>, July 1998.

   [<a id="ref-29">29</a>] Braden, B., "Requirements for internet hosts - application and
        support", STD 3, <a href="./rfc1123">RFC 1123</a>, October 1989.

   [<a id="ref-30">30</a>] Palme, J., "Common internet message headers", <a href="./rfc2076">RFC 2076</a>, February
        1997.

   [<a id="ref-31">31</a>] Alvestrand, H., "IETF policy on character sets and languages",
        <a href="./rfc2277">RFC 2277</a>, January 1998.

   [<a id="ref-32">32</a>] Elkins, M., "MIME security with pretty good privacy (PGP)", <a href="./rfc2015">RFC</a>
        <a href="./rfc2015">2015</a>, October 1996.

   [<a id="ref-33">33</a>] Atkins, D., Stallings, W. and P. Zimmermann, "PGP message
        exchange formats", <a href="./rfc1991">RFC 1991</a>, August 1996.

   [<a id="ref-34">34</a>] Atkinson, R., "Security architecture for the internet protocol",
        <a href="./rfc2401">RFC 2401</a>, November 1998.

   [<a id="ref-35">35</a>] Allen, C. and T. Dierks, "The TLS protocol version 1.0," <a href="./rfc2246">RFC</a>
        <a href="./rfc2246">2246</a>, January 1999.

   [<a id="ref-36">36</a>] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
        Leach, P., Luotonen, A. and L. Stewart, "HTTP authentication:
        Basic and digest access authentication," Internet Draft,
        Internet Engineering Task Force, Sept.  1998.  Work in progress.

   [<a id="ref-37">37</a>] Schooler, E., "Case study: multimedia conference control in a
        packet-switched teleconferencing system," Journal of
        Internetworking:  Research and Experience , vol. 4, pp. 99--120,
        June 1993.  ISI reprint series ISI/RS-93-359.

   [<a id="ref-38">38</a>] Schulzrinne, H., "Personal mobility for multimedia services in
        the Internet," in European Workshop on Interactive Distributed
        Multimedia Systems and Services (IDMS) , (Berlin, Germany), Mar.
        1996.













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Full Copyright Statement

   Copyright (C) The Internet Society (1999).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
























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